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// Copyright 2020, Chef. All rights reserved.
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// https://github.com/q191201771/lal
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//
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// Use of this source code is governed by a MIT-style license
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// that can be found in the License file.
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//
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// Author: Chef (191201771@qq.com)
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package rtsp
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import (
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"encoding/hex"
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"net"
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"sync"
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"time"
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"github.com/q191201771/naza/pkg/nazaatomic"
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"github.com/q191201771/naza/pkg/nazabytes"
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"github.com/q191201771/naza/pkg/nazaerrors"
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"github.com/q191201771/lal/pkg/base"
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"github.com/q191201771/lal/pkg/rtprtcp"
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"github.com/q191201771/lal/pkg/sdp"
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"github.com/q191201771/naza/pkg/connection"
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"github.com/q191201771/naza/pkg/nazalog"
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"github.com/q191201771/naza/pkg/nazanet"
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)
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// 聚合PubSession和PullSession，也即流数据是输入类型的session
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// BaseInSession会向上层回调两种格式的数据(本质上是一份数据，业务方可自由选择使用)：
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// 1. 原始的rtp packet
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// 2. rtp合并后的av packet
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//
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type BaseInSessionObserver interface {
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OnSdp(sdpCtx sdp.LogicContext)
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// 回调收到的RTP包
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OnRtpPacket(pkt rtprtcp.RtpPacket)
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// @param pkt: pkt结构体中字段含义见rtprtcp.OnAvPacket
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OnAvPacket(pkt base.AvPacket)
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}
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type BaseInSession struct {
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uniqueKey string // 使用上层Session的值
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cmdSession IInterleavedPacketWriter
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observer BaseInSessionObserver
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audioRtpConn *nazanet.UdpConnection
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videoRtpConn *nazanet.UdpConnection
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audioRtcpConn *nazanet.UdpConnection
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videoRtcpConn *nazanet.UdpConnection
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audioRtpChannel int
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audioRtcpChannel int
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videoRtpChannel int
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videoRtcpChannel int
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currConnStat connection.StatAtomic
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prevConnStat connection.Stat
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staleStat *connection.Stat
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stat base.StatSession
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mu sync.Mutex
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sdpCtx sdp.LogicContext // const after set
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avPacketQueue *AvPacketQueue
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audioRrProducer *rtprtcp.RrProducer
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videoRrProducer *rtprtcp.RrProducer
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audioUnpacker rtprtcp.IRtpUnpacker
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videoUnpacker rtprtcp.IRtpUnpacker
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audioSsrc nazaatomic.Uint32
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videoSsrc nazaatomic.Uint32
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// only for debug log
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debugLogMaxCount uint32
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loggedReadAudioRtpCount nazaatomic.Uint32
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loggedReadVideoRtpCount nazaatomic.Uint32
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loggedReadRtcpCount nazaatomic.Uint32
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loggedReadSrCount nazaatomic.Uint32
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disposeOnce sync.Once
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waitChan chan error
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}
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func NewBaseInSession(uniqueKey string, cmdSession IInterleavedPacketWriter) *BaseInSession {
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s := &BaseInSession{
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uniqueKey: uniqueKey,
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stat: base.StatSession{
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Protocol: base.ProtocolRtsp,
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SessionId: uniqueKey,
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StartTime: time.Now().Format("2006-01-02 15:04:05.999"),
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},
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cmdSession: cmdSession,
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debugLogMaxCount: 3,
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waitChan: make(chan error, 1),
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}
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nazalog.Infof("[%s] lifecycle new rtsp BaseInSession. session=%p", uniqueKey, s)
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return s
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}
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func NewBaseInSessionWithObserver(uniqueKey string, cmdSession IInterleavedPacketWriter, observer BaseInSessionObserver) *BaseInSession {
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s := NewBaseInSession(uniqueKey, cmdSession)
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s.observer = observer
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return s
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}
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func (session *BaseInSession) InitWithSdp(sdpCtx sdp.LogicContext) {
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session.mu.Lock()
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session.sdpCtx = sdpCtx
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session.mu.Unlock()
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if session.sdpCtx.IsAudioUnpackable() {
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session.audioUnpacker = rtprtcp.DefaultRtpUnpackerFactory(session.sdpCtx.GetAudioPayloadTypeBase(), session.sdpCtx.AudioClockRate, unpackerItemMaxSize, session.onAvPacketUnpacked)
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} else {
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nazalog.Warnf("[%s] audio unpacker not support for this type yet. logicCtx=%+v", session.uniqueKey, session.sdpCtx)
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}
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if session.sdpCtx.IsVideoUnpackable() {
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session.videoUnpacker = rtprtcp.DefaultRtpUnpackerFactory(session.sdpCtx.GetVideoPayloadTypeBase(), session.sdpCtx.VideoClockRate, unpackerItemMaxSize, session.onAvPacketUnpacked)
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} else {
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nazalog.Warnf("[%s] video unpacker not support this type yet. logicCtx=%+v", session.uniqueKey, session.sdpCtx)
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}
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session.audioRrProducer = rtprtcp.NewRrProducer(session.sdpCtx.AudioClockRate)
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session.videoRrProducer = rtprtcp.NewRrProducer(session.sdpCtx.VideoClockRate)
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if session.sdpCtx.IsAudioUnpackable() && session.sdpCtx.IsVideoUnpackable() {
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session.mu.Lock()
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session.avPacketQueue = NewAvPacketQueue(session.onAvPacket)
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session.mu.Unlock()
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}
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if session.observer != nil {
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session.observer.OnSdp(session.sdpCtx)
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}
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}
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// 如果没有设置回调监听对象，可以通过该函数设置，调用方保证调用该函数发生在调用InitWithSdp之后
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func (session *BaseInSession) SetObserver(observer BaseInSessionObserver) {
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session.observer = observer
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// 避免在当前协程回调，降低业务方使用负担，不必担心设置监听对象和回调函数中锁重入 TODO(chef): 更好的方式
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go func() {
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session.observer.OnSdp(session.sdpCtx)
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}()
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}
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func (session *BaseInSession) SetupWithConn(uri string, rtpConn, rtcpConn *nazanet.UdpConnection) error {
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if session.sdpCtx.IsAudioUri(uri) {
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session.audioRtpConn = rtpConn
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session.audioRtcpConn = rtcpConn
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} else if session.sdpCtx.IsVideoUri(uri) {
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session.videoRtpConn = rtpConn
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session.videoRtcpConn = rtcpConn
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} else {
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return ErrRtsp
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}
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go rtpConn.RunLoop(session.onReadRtpPacket)
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go rtcpConn.RunLoop(session.onReadRtcpPacket)
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return nil
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}
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func (session *BaseInSession) SetupWithChannel(uri string, rtpChannel, rtcpChannel int) error {
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if session.sdpCtx.IsAudioUri(uri) {
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session.audioRtpChannel = rtpChannel
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session.audioRtcpChannel = rtcpChannel
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return nil
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} else if session.sdpCtx.IsVideoUri(uri) {
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session.videoRtpChannel = rtpChannel
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session.videoRtcpChannel = rtcpChannel
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return nil
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}
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return ErrRtsp
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}
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// ---------------------------------------------------------------------------------------------------------------------
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// IClientSessionLifecycle interface
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// ---------------------------------------------------------------------------------------------------------------------
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// Dispose 文档请参考： IClientSessionLifecycle interface
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//
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func (session *BaseInSession) Dispose() error {
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return session.dispose(nil)
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}
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// WaitChan 文档请参考： IClientSessionLifecycle interface
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//
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// 注意，目前只有一种情况，即上层主动调用Dispose函数，此时error为nil
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//
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func (session *BaseInSession) WaitChan() <-chan error {
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return session.waitChan
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}
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// ---------------------------------------------------------------------------------------------------------------------
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func (session *BaseInSession) GetSdp() sdp.LogicContext {
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session.mu.Lock()
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defer session.mu.Unlock()
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return session.sdpCtx
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}
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func (session *BaseInSession) HandleInterleavedPacket(b []byte, channel int) {
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switch channel {
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case session.audioRtpChannel:
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fallthrough
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case session.videoRtpChannel:
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_ = session.handleRtpPacket(b)
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case session.audioRtcpChannel:
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fallthrough
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case session.videoRtcpChannel:
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_ = session.handleRtcpPacket(b, nil)
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default:
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nazalog.Errorf("[%s] read interleaved packet but channel invalid. channel=%d", session.uniqueKey, channel)
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}
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}
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// 发现pull时，需要先给对端发送数据，才能收到数据
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func (session *BaseInSession) WriteRtpRtcpDummy() {
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if session.videoRtpConn != nil {
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_ = session.videoRtpConn.Write(dummyRtpPacket)
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}
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if session.videoRtcpConn != nil {
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_ = session.videoRtcpConn.Write(dummyRtcpPacket)
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}
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if session.audioRtpConn != nil {
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_ = session.audioRtpConn.Write(dummyRtpPacket)
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}
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if session.audioRtcpConn != nil {
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_ = session.audioRtcpConn.Write(dummyRtcpPacket)
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}
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}
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func (session *BaseInSession) GetStat() base.StatSession {
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session.stat.ReadBytesSum = session.currConnStat.ReadBytesSum.Load()
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session.stat.WroteBytesSum = session.currConnStat.WroteBytesSum.Load()
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return session.stat
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}
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func (session *BaseInSession) UpdateStat(intervalSec uint32) {
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readBytesSum := session.currConnStat.ReadBytesSum.Load()
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wroteBytesSum := session.currConnStat.WroteBytesSum.Load()
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rDiff := readBytesSum - session.prevConnStat.ReadBytesSum
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session.stat.ReadBitrate = int(rDiff * 8 / 1024 / uint64(intervalSec))
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wDiff := wroteBytesSum - session.prevConnStat.WroteBytesSum
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session.stat.WriteBitrate = int(wDiff * 8 / 1024 / uint64(intervalSec))
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session.stat.Bitrate = session.stat.ReadBitrate
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session.prevConnStat.ReadBytesSum = readBytesSum
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session.prevConnStat.WroteBytesSum = wroteBytesSum
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}
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func (session *BaseInSession) IsAlive() (readAlive, writeAlive bool) {
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readBytesSum := session.currConnStat.ReadBytesSum.Load()
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wroteBytesSum := session.currConnStat.WroteBytesSum.Load()
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if session.staleStat == nil {
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session.staleStat = new(connection.Stat)
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session.staleStat.ReadBytesSum = readBytesSum
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session.staleStat.WroteBytesSum = wroteBytesSum
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return true, true
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}
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readAlive = !(readBytesSum-session.staleStat.ReadBytesSum == 0)
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writeAlive = !(wroteBytesSum-session.staleStat.WroteBytesSum == 0)
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session.staleStat.ReadBytesSum = readBytesSum
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session.staleStat.WroteBytesSum = wroteBytesSum
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return
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}
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func (session *BaseInSession) UniqueKey() string {
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return session.uniqueKey
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}
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// callback by RTPUnpacker
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func (session *BaseInSession) onAvPacketUnpacked(pkt base.AvPacket) {
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session.mu.Lock()
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defer session.mu.Unlock()
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if session.avPacketQueue != nil {
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session.avPacketQueue.Feed(pkt)
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} else {
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session.observer.OnAvPacket(pkt)
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}
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}
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// callback by avpacket queue
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func (session *BaseInSession) onAvPacket(pkt base.AvPacket) {
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session.observer.OnAvPacket(pkt)
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}
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// callback by UDPConnection
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func (session *BaseInSession) onReadRtpPacket(b []byte, rAddr *net.UDPAddr, err error) bool {
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if err != nil {
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// TODO(chef):
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// read udp [::]:30008: use of closed network connection
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// 可以退出loop，看是在上层退还是下层退，但是要注意每次read都判断的开销
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nazalog.Warnf("[%s] read udp packet failed. err=%+v", session.uniqueKey, err)
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return true
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}
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_ = session.handleRtpPacket(b)
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return true
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}
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// callback by UDPConnection
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func (session *BaseInSession) onReadRtcpPacket(b []byte, rAddr *net.UDPAddr, err error) bool {
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if err != nil {
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nazalog.Warnf("[%s] read udp packet failed. err=%+v", session.uniqueKey, err)
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return true
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}
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_ = session.handleRtcpPacket(b, rAddr)
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return true
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}
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// @param rAddr 对端地址，往对端发送数据时使用，注意，如果nil，则表示是interleaved模式，我们直接往TCP连接发数据
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func (session *BaseInSession) handleRtcpPacket(b []byte, rAddr *net.UDPAddr) error {
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session.currConnStat.ReadBytesSum.Add(uint64(len(b)))
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if len(b) <= 0 {
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nazalog.Errorf("[%s] handleRtcpPacket but length invalid. len=%d", session.uniqueKey, len(b))
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return ErrRtsp
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}
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if session.loggedReadRtcpCount.Load() < session.debugLogMaxCount {
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nazalog.Debugf("[%s] LOGPACKET. read rtcp=%s", session.uniqueKey, hex.Dump(nazabytes.Prefix(b, 32)))
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session.loggedReadRtcpCount.Increment()
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}
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packetType := b[1]
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switch packetType {
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case rtprtcp.RtcpPacketTypeSr:
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sr := rtprtcp.ParseSr(b)
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if session.loggedReadSrCount.Load() < session.debugLogMaxCount {
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nazalog.Debugf("[%s] LOGPACKET. %+v", session.uniqueKey, sr)
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session.loggedReadSrCount.Increment()
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}
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var rrBuf []byte
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switch sr.SenderSsrc {
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case session.audioSsrc.Load():
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session.mu.Lock()
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rrBuf = session.audioRrProducer.Produce(sr.GetMiddleNtp())
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session.mu.Unlock()
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if rrBuf != nil {
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if rAddr != nil {
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_ = session.audioRtcpConn.Write2Addr(rrBuf, rAddr)
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} else {
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_ = session.cmdSession.WriteInterleavedPacket(rrBuf, session.audioRtcpChannel)
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}
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session.currConnStat.WroteBytesSum.Add(uint64(len(b)))
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}
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case session.videoSsrc.Load():
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session.mu.Lock()
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rrBuf = session.videoRrProducer.Produce(sr.GetMiddleNtp())
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session.mu.Unlock()
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if rrBuf != nil {
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if rAddr != nil {
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_ = session.videoRtcpConn.Write2Addr(rrBuf, rAddr)
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} else {
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_ = session.cmdSession.WriteInterleavedPacket(rrBuf, session.videoRtcpChannel)
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}
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session.currConnStat.WroteBytesSum.Add(uint64(len(b)))
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}
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default:
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// ffmpeg推流时，会在发送第一个RTP包之前就发送一个SR，所以关闭这个警告日志
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//nazalog.Warnf("[%s] read rtcp sr but senderSsrc invalid. senderSsrc=%d, audio=%d, video=%d",
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// p.uniqueKey, sr.SenderSsrc, p.audioSsrc, p.videoSsrc)
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return ErrRtsp
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}
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default:
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nazalog.Warnf("[%s] handleRtcpPacket but type unknown. type=%d", session.uniqueKey, b[1])
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return ErrRtsp
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}
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return nil
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}
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func (session *BaseInSession) handleRtpPacket(b []byte) error {
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session.currConnStat.ReadBytesSum.Add(uint64(len(b)))
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if len(b) < rtprtcp.RtpFixedHeaderLength {
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nazalog.Errorf("[%s] handleRtpPacket but length invalid. len=%d", session.uniqueKey, len(b))
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return ErrRtsp
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}
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packetType := int(b[1] & 0x7F)
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if !session.sdpCtx.IsPayloadTypeOrigin(packetType) {
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nazalog.Errorf("[%s] handleRtpPacket but type invalid. type=%d", session.uniqueKey, packetType)
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return ErrRtsp
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}
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h, err := rtprtcp.ParseRtpHeader(b)
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if err != nil {
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|
nazalog.Errorf("[%s] handleRtpPacket invalid rtp packet. err=%+v", session.uniqueKey, err)
|
|
|
return err
|
|
|
}
|
|
|
|
|
|
var pkt rtprtcp.RtpPacket
|
|
|
pkt.Header = h
|
|
|
pkt.Raw = b
|
|
|
|
|
|
// 接收数据时，保证了sdp的原始类型对应
|
|
|
if session.sdpCtx.IsAudioPayloadTypeOrigin(packetType) {
|
|
|
if session.loggedReadAudioRtpCount.Load() < session.debugLogMaxCount {
|
|
|
nazalog.Debugf("[%s] LOGPACKET. read audio rtp=%+v, len=%d", session.uniqueKey, h, len(b))
|
|
|
session.loggedReadAudioRtpCount.Increment()
|
|
|
}
|
|
|
|
|
|
session.audioSsrc.Store(h.Ssrc)
|
|
|
session.observer.OnRtpPacket(pkt)
|
|
|
session.mu.Lock()
|
|
|
session.audioRrProducer.FeedRtpPacket(h.Seq)
|
|
|
session.mu.Unlock()
|
|
|
|
|
|
if session.audioUnpacker != nil {
|
|
|
session.audioUnpacker.Feed(pkt)
|
|
|
}
|
|
|
} else if session.sdpCtx.IsVideoPayloadTypeOrigin(packetType) {
|
|
|
if session.loggedReadVideoRtpCount.Load() < session.debugLogMaxCount {
|
|
|
nazalog.Debugf("[%s] LOGPACKET. read video rtp=%+v, len=%d", session.uniqueKey, h, len(b))
|
|
|
session.loggedReadVideoRtpCount.Increment()
|
|
|
}
|
|
|
|
|
|
session.videoSsrc.Store(h.Ssrc)
|
|
|
session.observer.OnRtpPacket(pkt)
|
|
|
session.mu.Lock()
|
|
|
session.videoRrProducer.FeedRtpPacket(h.Seq)
|
|
|
session.mu.Unlock()
|
|
|
|
|
|
if session.videoUnpacker != nil {
|
|
|
session.videoUnpacker.Feed(pkt)
|
|
|
}
|
|
|
} else {
|
|
|
// noop 因为前面已经判断过type了，所以永远不会走到这
|
|
|
}
|
|
|
|
|
|
return nil
|
|
|
}
|
|
|
|
|
|
func (session *BaseInSession) dispose(err error) error {
|
|
|
var retErr error
|
|
|
session.disposeOnce.Do(func() {
|
|
|
nazalog.Infof("[%s] lifecycle dispose rtsp BaseInSession. session=%p", session.uniqueKey, session)
|
|
|
var e1, e2, e3, e4 error
|
|
|
if session.audioRtpConn != nil {
|
|
|
e1 = session.audioRtpConn.Dispose()
|
|
|
}
|
|
|
if session.audioRtcpConn != nil {
|
|
|
e2 = session.audioRtcpConn.Dispose()
|
|
|
}
|
|
|
if session.videoRtpConn != nil {
|
|
|
e3 = session.videoRtpConn.Dispose()
|
|
|
}
|
|
|
if session.videoRtcpConn != nil {
|
|
|
e4 = session.videoRtcpConn.Dispose()
|
|
|
}
|
|
|
|
|
|
session.waitChan <- nil
|
|
|
|
|
|
retErr = nazaerrors.CombineErrors(e1, e2, e3, e4)
|
|
|
})
|
|
|
return retErr
|
|
|
}
|