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[fix] 修复lalserver中（rtsp.BaseInSession以及logic.Group）的一些竞态读写，https://github.com/q191201771/lal/issues/47

Browse Source
pull/49/head
q191201771 4 years ago
parent 7c07f2ba05
commit 9bb3dac3d1
2 changed files with 55 additions and 40 deletions
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20
pkg/logic/group.go
Unescape Escape View File

@ -11,12 +11,11 @@ package logic
import (
"encoding/json"
"fmt"
"github.com/q191201771/lal/pkg/remux"
"os"
"strings"
"sync"
"github.com/q191201771/lal/pkg/remux"
"github.com/q191201771/naza/pkg/defertaskthread"
"github.com/q191201771/lal/pkg/rtprtcp"
@ -461,6 +460,12 @@ func (group *Group) HasOutSession() bool {
return group.hasOutSession()
}
func (group *Group) BroadcastRTMP(msg base.RTMPMsg) {
group.mutex.Lock()
defer group.mutex.Unlock()
group.broadcastRTMP(msg)
}
// hls.Muxer
func (group *Group) OnTSPackets(rawFrame []byte, boundary bool) {
// 因为最前面Feed时已经加锁了，所以这里回调上来就不用加锁了
@ -479,10 +484,7 @@ func (group *Group) OnTSPackets(rawFrame []byte, boundary bool) {
// rtmp.PubSession or rtmp.PullSession
func (group *Group) OnReadRTMPAVMsg(msg base.RTMPMsg) {
group.mutex.Lock()
defer group.mutex.Unlock()
group.broadcastRTMP(msg)
group.BroadcastRTMP(msg)
}
// rtsp.PubSession
@ -522,15 +524,13 @@ func (group *Group) OnAVConfig(asc, vps, sps, pps []byte) {
// rtsp.PubSession
func (group *Group) OnAVPacket(pkt base.AVPacket) {
// TODO chef: 这里没有加锁，最起码下面广播前需要加锁
msg, err := remux.AVPacket2RTMPMsg(pkt)
if err != nil {
nazalog.Errorf("[%s] remux av packet to rtmp msg failed. err=+%v", group.UniqueKey, err)
return
}
group.broadcastRTMP(msg)
group.BroadcastRTMP(msg)
}
func (group *Group) StringifyDebugStats() string {
@ -1012,7 +1012,7 @@ func (group *Group) disposeHLSMuxer() {
streamName := param[1].(string)
outPath := param[2].(string)
if g := sm.getGroup(appName, streamName); g != nil {
if g := sm.GetGroup(appName, streamName); g != nil {
if g.IsHLSMuxerAlive() {
nazalog.Warnf("cancel cleanup hls file path since hls muxer still alive. streamName=%s", streamName)
return

75
pkg/rtsp/base_in_session.go
Unescape Escape View File

@ -10,6 +10,7 @@ package rtsp
import (
"encoding/hex"
"github.com/q191201771/naza/pkg/nazaatomic"
"net"
"sync"
"time"
@ -63,24 +64,25 @@ type BaseInSession struct {
staleStat *connection.Stat
stat base.StatSession
m sync.Mutex
mu sync.Mutex
rawSDP []byte // const after set
sdpLogicCtx sdp.LogicContext // const after set
avPacketQueue *AVPacketQueue
audioUnpacker *rtprtcp.RTPUnpacker
videoUnpacker *rtprtcp.RTPUnpacker
audioRRProducer *rtprtcp.RRProducer
videoRRProducer *rtprtcp.RRProducer
audioSSRC uint32
videoSSRC uint32
audioUnpacker *rtprtcp.RTPUnpacker
videoUnpacker *rtprtcp.RTPUnpacker
audioSSRC nazaatomic.Uint32
videoSSRC nazaatomic.Uint32
// only for debug log
debugLogMaxCount int
loggedReadAudioRTPCount int
loggedReadVideoRTPCount int
loggedReadRTCPCount int
loggedReadSRCount int
debugLogMaxCount uint32
loggedReadAudioRTPCount nazaatomic.Uint32
loggedReadVideoRTPCount nazaatomic.Uint32
loggedReadRTCPCount nazaatomic.Uint32
loggedReadSRCount nazaatomic.Uint32
}
func NewBaseInSession(uniqueKey string, cmdSession IInterleavedPacketWriter) *BaseInSession {
@ -105,10 +107,10 @@ func NewBaseInSessionWithObserver(uniqueKey string, cmdSession IInterleavedPacke
}
func (session *BaseInSession) InitWithSDP(rawSDP []byte, sdpLogicCtx sdp.LogicContext) {
session.m.Lock()
session.mu.Lock()
session.rawSDP = rawSDP
session.sdpLogicCtx = sdpLogicCtx
session.m.Unlock()
session.mu.Unlock()
if session.sdpLogicCtx.IsAudioUnpackable() {
session.audioUnpacker = rtprtcp.NewRTPUnpacker(session.sdpLogicCtx.GetAudioPayloadTypeBase(), session.sdpLogicCtx.AudioClockRate, unpackerItemMaxSize, session.onAVPacketUnpacked)
@ -125,7 +127,9 @@ func (session *BaseInSession) InitWithSDP(rawSDP []byte, sdpLogicCtx sdp.LogicCo
session.videoRRProducer = rtprtcp.NewRRProducer(session.sdpLogicCtx.VideoClockRate)
if session.sdpLogicCtx.IsAudioUnpackable() && session.sdpLogicCtx.IsVideoUnpackable() {
session.mu.Lock()
session.avPacketQueue = NewAVPacketQueue(session.onAVPacket)
session.mu.Unlock()
}
if session.observer != nil {
@ -189,8 +193,8 @@ func (session *BaseInSession) Dispose() error {
}
func (session *BaseInSession) GetSDP() ([]byte, sdp.LogicContext) {
session.m.Lock()
defer session.m.Unlock()
session.mu.Lock()
defer session.mu.Unlock()
return session.rawSDP, session.sdpLogicCtx
}
@ -266,6 +270,9 @@ func (session *BaseInSession) UniqueKey() string {
// callback by RTPUnpacker
func (session *BaseInSession) onAVPacketUnpacked(pkt base.AVPacket) {
session.mu.Lock()
defer session.mu.Unlock()
if session.avPacketQueue != nil {
session.avPacketQueue.Feed(pkt)
} else {
@ -309,9 +316,9 @@ func (session *BaseInSession) handleRTCPPacket(b []byte, rAddr *net.UDPAddr) err
return ErrRTSP
}
if session.loggedReadRTCPCount < session.debugLogMaxCount {
if session.loggedReadRTCPCount.Load() < session.debugLogMaxCount {
nazalog.Debugf("[%s] LOGPACKET. read rtcp=%s", session.uniqueKey, hex.Dump(nazastring.SubSliceSafety(b, 32)))
session.loggedReadRTCPCount++
session.loggedReadRTCPCount.Increment()
}
packetType := b[1]
@ -319,14 +326,16 @@ func (session *BaseInSession) handleRTCPPacket(b []byte, rAddr *net.UDPAddr) err
switch packetType {
case rtprtcp.RTCPPacketTypeSR:
sr := rtprtcp.ParseSR(b)
if session.loggedReadSRCount < session.debugLogMaxCount {
if session.loggedReadSRCount.Load() < session.debugLogMaxCount {
nazalog.Debugf("[%s] LOGPACKET. %+v", session.uniqueKey, sr)
session.loggedReadSRCount++
session.loggedReadSRCount.Increment()
}
var rrBuf []byte
switch sr.SenderSSRC {
case session.audioSSRC:
case session.audioSSRC.Load():
session.mu.Lock()
rrBuf = session.audioRRProducer.Produce(sr.GetMiddleNTP())
session.mu.Unlock()
if rrBuf != nil {
if rAddr != nil {
_ = session.audioRTCPConn.Write2Addr(rrBuf, rAddr)
@ -335,8 +344,10 @@ func (session *BaseInSession) handleRTCPPacket(b []byte, rAddr *net.UDPAddr) err
}
session.currConnStat.WroteBytesSum.Add(uint64(len(b)))
}
case session.videoSSRC:
case session.videoSSRC.Load():
session.mu.Lock()
rrBuf = session.videoRRProducer.Produce(sr.GetMiddleNTP())
session.mu.Unlock()
if rrBuf != nil {
if rAddr != nil {
_ = session.videoRTCPConn.Write2Addr(rrBuf, rAddr)
@ -373,7 +384,7 @@ func (session *BaseInSession) handleRTPPacket(b []byte) error {
return ErrRTSP
}
h, err := rtprtcp.ParseRTPPacket(b)
h, err := rtprtcp.ParseRTPHeader(b)
if err != nil {
nazalog.Errorf("[%s] handleRTPPacket invalid rtp packet. err=%+v", session.uniqueKey, err)
return err
@ -385,27 +396,31 @@ func (session *BaseInSession) handleRTPPacket(b []byte) error {
// 接收数据时，保证了sdp的原始类型对应
if session.sdpLogicCtx.IsAudioPayloadTypeOrigin(packetType) {
if session.loggedReadAudioRTPCount < session.debugLogMaxCount {
nazalog.Debugf("[%s] LOGPACKET. read audio rtp=%+v", session.uniqueKey, h)
session.loggedReadAudioRTPCount++
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 = h.SSRC
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.sdpLogicCtx.IsVideoPayloadTypeOrigin(packetType) {
if session.loggedReadVideoRTPCount < session.debugLogMaxCount {
nazalog.Debugf("[%s] LOGPACKET. read video rtp=%+v", session.uniqueKey, h)
session.loggedReadVideoRTPCount++
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 = h.SSRC
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)

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