apps/plugin-webrtc
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开发播放功能

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This commit is contained in:
langhuihui
2020-07-01 23:49:13 +08:00
parent 4922863211
commit 9990b2b536
1 changed files with 32 additions and 29 deletions
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61
main.go
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@@ -6,6 +6,7 @@ import (
"fmt"
"io/ioutil"
"net/http"
"os"
"sync"
"time"
@@ -74,6 +75,26 @@ type WebRTC struct {
*PeerConnection
RemoteAddr string
videoTrack *Track
sequence uint16
codecs.H264Packet
*os.File
}
func (rtc *WebRTC) WriteVideo(ts uint32, marker bool, payload []byte) error {
rtc.sequence++
// bb, _ := rtc.Unmarshal(payload)
// rtc.Write(bb)
return rtc.videoTrack.WriteRTP(&rtp.Packet{
Header: rtp.Header{
Version: 2,
SSRC: SSRC,
PayloadType: DefaultPayloadTypeH264,
SequenceNumber: rtc.sequence,
Timestamp: ts,
Marker: marker,
},
Payload: payload,
})
}
func (rtc *WebRTC) Play(streamPath string) bool {
@@ -85,26 +106,14 @@ func (rtc *WebRTC) Play(streamPath string) bool {
rtc.Stream.Close()
}
case ICEConnectionStateConnected:
var sequence uint16
var sub Subscriber
var sps []byte
var pps []byte
sub.ID = rtc.RemoteAddr
sub.Type = "WebRTC"
nextHeader := func(ts uint32, marker bool) rtp.Header {
sequence++
return rtp.Header{
Version: 2,
SSRC: SSRC,
PayloadType: DefaultPayloadTypeH264,
SequenceNumber: sequence,
Timestamp: ts,
Marker: marker,
}
}
stapA := func(naul ...[]byte) []byte {
var buffer bytes.Buffer
buffer.WriteByte(24)
buffer.WriteByte((naul[0][0] & 224) | 24)
for _, n := range naul {
l := len(n)
buffer.WriteByte(byte(l >> 8))
@@ -113,8 +122,8 @@ func (rtc *WebRTC) Play(streamPath string) bool {
}
return buffer.Bytes()
}
// aud := []byte{0x09, 0x30}
//rtc.File, _ = os.OpenFile("webrtc.h264", os.O_CREATE|os.O_WRONLY|os.O_TRUNC, 0666)
aud := []byte{0x09, 0x30}
sub.OnData = func(packet *avformat.SendPacket) error {
if packet.Type == avformat.FLV_TAG_TYPE_AUDIO {
return nil
@@ -128,10 +137,11 @@ func (rtc *WebRTC) Play(streamPath string) bool {
pps = payload[2:ppsLen]
} else {
if packet.IsKeyFrame {
if err := rtc.videoTrack.WriteRTP(&rtp.Packet{
Header: nextHeader(packet.Timestamp*90, true),
Payload: stapA(sps, pps),
}); err != nil {
if err := rtc.WriteVideo(packet.Timestamp*90, true, stapA([]byte{0x9, 0x10}, sps, pps)); err != nil {
return err
}
} else {
if err := rtc.WriteVideo(packet.Timestamp*90, true, aud); err != nil {
return err
}
}
@@ -147,10 +157,7 @@ func (rtc *WebRTC) Play(streamPath string) bool {
part := _payload[1:1000]
marker := false
for {
if err := rtc.videoTrack.WriteRTP(&rtp.Packet{
Header: nextHeader(packet.Timestamp*90, marker),
Payload: append([]byte{indicator, header}, part...),
}); err != nil {
if err := rtc.WriteVideo(packet.Timestamp*90, marker, append([]byte{indicator, header}, part...)); err != nil {
return err
}
if _payload == nil {
@@ -168,10 +175,7 @@ func (rtc *WebRTC) Play(streamPath string) bool {
}
}
} else {
if err := rtc.videoTrack.WriteRTP(&rtp.Packet{
Header: nextHeader(packet.Timestamp*90, true),
Payload: _payload,
}); err != nil {
if err := rtc.WriteVideo(packet.Timestamp*90, true, _payload); err != nil {
return err
}
}
@@ -223,7 +227,6 @@ func (rtc *WebRTC) Publish(streamPath string) bool {
rtc.PeerConnection = peerConnection
if rtc.RTP.Publish(streamPath) {
//f, _ := os.OpenFile("resource/live/rtc.h264", os.O_TRUNC|os.O_WRONLY, 0666)
var h264 codecs.H264Packet
rtc.Stream.Type = "WebRTC"
peerConnection.OnTrack(func(track *Track, receiver *RTPReceiver) {
defer rtc.Stream.Close()
@@ -249,7 +252,7 @@ func (rtc *WebRTC) Publish(streamPath string) bool {
if err = pack.Unmarshal(b[:i]); err != nil {
return
}
h264.Unmarshal(pack.Payload)
rtc.Unmarshal(pack.Payload)
// f.Write(bytes)
}
})