apps/plugin-rtsp
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适配引擎版本升级

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This commit is contained in:
langhuihui
2021-06-15 08:06:40 +08:00
parent d08230bf0c
commit b0c3cdb21a
6 changed files with 92 additions and 165 deletions
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37
client.go
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@@ -18,13 +18,15 @@ import (
. "github.com/Monibuca/engine/v3"
. "github.com/Monibuca/utils/v3"
"github.com/pion/rtp"
)
// PullStream 从外部拉流
func (rtsp *RTSP) PullStream(streamPath string, rtspUrl string) (err error) {
if result := rtsp.Publish(streamPath); result {
rtsp.Stream.Type = "RTSP"
rtsp.Stream = &Stream{
StreamPath: streamPath,
Type: "RTSP Pull",
}
if result := rtsp.Publish(); result {
rtsp.TransType = TRANS_TYPE_TCP
rtsp.vRTPChannel = 0
rtsp.vRTPControlChannel = 1
@@ -217,14 +219,12 @@ func (client *RTSP) requestStream() (err error) {
switch t {
case "video":
if len(sdpInfo.SpropParameterSets) > 1 {
vt := NewVideoTrack()
vt.CodecID = 7
var pack VideoPack
pack.Payload = sdpInfo.SpropParameterSets[0]
vt.Push(pack)
pack.Payload = sdpInfo.SpropParameterSets[1]
vt.Push(pack)
client.SetOriginVT(vt)
client.RtpVideo = client.NewRTPVideo(7)
client.RtpVideo.PushNalu(VideoPack{NALUs: sdpInfo.SpropParameterSets})
} else if client.VSdp.Codec == "H264" {
client.RtpVideo = client.NewRTPVideo(7)
} else if client.VSdp.Codec == "H265" {
client.RtpVideo = client.NewRTPVideo(12)
}
if client.TransType == TRANS_TYPE_TCP {
headers["Transport"] = fmt.Sprintf("RTP/AVP/TCP;unicast;interleaved=%d-%d", client.vRTPChannel, client.vRTPControlChannel)
@@ -238,13 +238,13 @@ func (client *RTSP) requestStream() (err error) {
client.Conn.timeout = 0 // UDP ignore timeout
}
case "audio":
at := NewAudioTrack()
if len(sdpInfo.Config) > 0 {
at.SetASC(sdpInfo.Config)
client.RtpAudio = client.NewRTPAudio(0)
at := client.RtpAudio.AudioTrack
if len(client.ASdp.Control) > 0 {
at.SetASC(client.ASdp.Config)
} else {
client.setAudioFormat(at)
}
client.SetOriginAT(at)
if client.TransType == TRANS_TYPE_TCP {
headers["Transport"] = fmt.Sprintf("RTP/AVP/TCP;unicast;interleaved=%d-%d", client.aRTPChannel, client.aRTPControlChannel)
} else {
@@ -339,15 +339,14 @@ func (client *RTSP) startStream() {
Printf("io.ReadFull err:%v", err)
return
}
var pack rtp.Packet
pack.Unmarshal(content)
switch channel {
case client.aRTPChannel:
client.OriginAudioTrack.PushRTP(pack)
client.RtpAudio.Push(content)
case client.aRTPControlChannel:
case client.vRTPChannel:
client.OriginVideoTrack.PushRTP(pack)
client.RtpVideo.Push(content)
case client.vRTPControlChannel:
default: