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fix audio del 4 byte on start fix webrtc add options ice and port, webrtc.MediaEngine{}

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This commit is contained in:
Andrey Semochkin
2021-01-15 18:26:23 +03:00
parent f00d1c189a
commit 8d167fd1c0
3 changed files with 59 additions and 22 deletions
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25
format/rtspv2/client.go
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@@ -522,7 +522,6 @@ func (client *RTSPClient) RTPDemuxer(payloadRAW *[]byte) ([]*av.Packet, bool) {
client.BufferRtpPacket.Truncate(0)
client.BufferRtpPacket.Reset()
}
nalRaw, _ := h264parser.SplitNALUs(content[offset:end])
var retmap []*av.Packet
for _, nal := range nalRaw {
@@ -559,6 +558,22 @@ func (client *RTSPClient) RTPDemuxer(payloadRAW *[]byte) ([]*av.Packet, bool) {
if isEnd {
client.fuStarted = false
naluTypef := client.BufferRtpPacket.Bytes()[0] & 0x1f
if naluTypef == 7 {
bufered, _ := h264parser.SplitNALUs(append([]byte{0, 0, 0, 1}, client.BufferRtpPacket.Bytes()...))
for _, v := range bufered {
naluTypefs := v[0] & 0x1f
switch {
case naluTypefs == 5:
client.BufferRtpPacket.Reset()
client.BufferRtpPacket.Write(v)
naluTypef = 5
case naluTypefs == 7:
client.CodecUpdateSPS(v)
case naluTypefs == 8:
client.CodecUpdatePPS(v)
}
}
}
retmap = append(retmap, &av.Packet{
Data: append(binSize(client.BufferRtpPacket.Len()), client.BufferRtpPacket.Bytes()...),
CompositionTime: time.Duration(1) * time.Millisecond,
@@ -591,7 +606,7 @@ func (client *RTSPClient) RTPDemuxer(payloadRAW *[]byte) ([]*av.Packet, bool) {
duration = time.Duration(len(nal)) * time.Second / time.Duration(client.AudioTimeScale)
client.AudioTimeLine += duration
retmap = append(retmap, &av.Packet{
Data: append(binSize(len(nal)), nal...),
Data: nal,
CompositionTime: time.Duration(1) * time.Millisecond,
Duration: duration,
Idx: client.audioIDX,
@@ -602,7 +617,7 @@ func (client *RTSPClient) RTPDemuxer(payloadRAW *[]byte) ([]*av.Packet, bool) {
duration = time.Duration(len(nal)) * time.Second / time.Duration(client.AudioTimeScale)
client.AudioTimeLine += duration
retmap = append(retmap, &av.Packet{
Data: append(binSize(len(nal)), nal...),
Data: nal,
CompositionTime: time.Duration(1) * time.Millisecond,
Duration: duration,
Idx: client.audioIDX,
@@ -613,7 +628,7 @@ func (client *RTSPClient) RTPDemuxer(payloadRAW *[]byte) ([]*av.Packet, bool) {
duration = time.Duration(20) * time.Millisecond
client.AudioTimeLine += duration
retmap = append(retmap, &av.Packet{
Data: append(binSize(len(nal)), nal...),
Data: nal,
CompositionTime: time.Duration(1) * time.Millisecond,
Duration: duration,
Idx: client.audioIDX,
@@ -638,7 +653,7 @@ func (client *RTSPClient) RTPDemuxer(payloadRAW *[]byte) ([]*av.Packet, bool) {
duration = time.Duration((float32(1024)/float32(client.AudioTimeScale))*1000) * time.Millisecond
client.AudioTimeLine += duration
retmap = append(retmap, &av.Packet{
Data: append(binSize(len(frame)), frame...),
Data: frame,
CompositionTime: time.Duration(1) * time.Millisecond,
Duration: duration,
Idx: client.audioIDX,