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opus work worked

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This commit is contained in:
Andrey Semochkin 2021-01-09 17:24:48 +03:00
parent 6a92c34cd4
commit 33b07c6a20
2 changed files with 19 additions and 10 deletions
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21
format/rtspv2/client.go
View File

@ -75,8 +75,9 @@ type RTSPClient struct {
sps []byte
pps []byte
CodecData []av.CodecData
PCMTime time.Duration
AudioTimeLine time.Duration
AudioTimeScale int64
audioCodec string
}
type RTSPClientOptions struct {
@ -179,6 +180,7 @@ func Dial(options RTSPClientOptions) (*RTSPClient, error) {
if CodecData != nil {
client.CodecData = append(client.CodecData, CodecData)
client.audioIDX = int8(len(client.CodecData) - 1)
client.audioCodec = CodecData.Type().String()
if i2.TimeScale != 0 {
client.AudioTimeScale = int64(i2.TimeScale)
}
@ -572,21 +574,20 @@ func (client *RTSPClient) RTPDemuxer(payloadRAW *[]byte) ([]*av.Packet, bool) {
nalRaw, _ := h264parser.SplitNALUs(content[offset:end])
var retmap []*av.Packet
for _, nal := range nalRaw {
//basic
//time.Duration(float32(timestamp)/float32(float32(client.AudioTimeScale)/float32(1000))) * time.Millisecond
//pcm
//client.PCMTime += time.Duration(len(nal)) * time.Second / time.Duration(client.AudioTimeScale)
//opus
//client.PCMTime := time.Duration((sampleCount/48000)*1000) * time.Millisecond
//Need Add Opus And AAC
if client.audioCodec == av.PCM_MULAW.String() || client.audioCodec == av.PCM_ALAW.String() || client.audioCodec == av.PCM.String() {
client.AudioTimeLine += time.Duration(len(nal)) * time.Second / time.Duration(client.AudioTimeScale)
} else if client.audioCodec == av.OPUS.String() {
client.AudioTimeLine += time.Duration(20) * time.Millisecond
} else {
client.AudioTimeLine = time.Duration(float32(timestamp)/float32(float32(client.AudioTimeScale)/float32(1000))) * time.Millisecond
}
retmap = append(retmap, &av.Packet{
Data: append(binSize(len(nal)), nal...),
CompositionTime: time.Duration(1) * time.Millisecond,
Idx: client.audioIDX,
IsKeyFrame: false,
Time: time.Duration(float32(timestamp)/float32(float32(client.AudioTimeScale)/float32(1000))) * time.Millisecond,
Time: client.AudioTimeLine,
})
//log.Println("===>", time.Duration(float32(timestamp)/float32(float32(client.AudioTimeScale)/float32(1000)))*time.Millisecond)
}
if len(retmap) > 0 {
return retmap, true

8
format/webrtcv3/adapter.go
View File

@ -159,6 +159,11 @@ func (element *Muxer) WritePacket(pkt av.Packet) (err error) {
if element.stop {
return ErrorClientOffline
}
//Wait client ICEConnectionStateConnected
if element.status == webrtc.ICEConnectionStateChecking {
WritePacketSuccess = true
return nil
}
if element.status != webrtc.ICEConnectionStateConnected {
return nil
}
@ -176,8 +181,11 @@ func (element *Muxer) WritePacket(pkt av.Packet) (err error) {
pkt.Data = pkt.Data[4:]
}
case av.PCM_MULAW:
pkt.Data = pkt.Data[4:]
case av.PCM_ALAW:
pkt.Data = pkt.Data[4:]
case av.OPUS:
pkt.Data = pkt.Data[4:]
default:
return ErrorCodecNotSupported
}