apps/plugin-webrtc
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增加播放功能

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This commit is contained in:
langhuihui
2020-07-05 21:43:44 +08:00
parent 457c8ff185
commit 854907f332
12 changed files with 549 additions and 479 deletions
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153
main.go
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@@ -1,12 +1,10 @@
package webrtc
import (
"bytes"
"encoding/json"
"fmt"
"io/ioutil"
"net/http"
"os"
"sync"
"time"
@@ -15,8 +13,6 @@ import (
"github.com/Monibuca/engine/v2/util"
. "github.com/Monibuca/plugin-rtp"
"github.com/pion/rtcp"
"github.com/pion/rtp"
"github.com/pion/rtp/codecs"
. "github.com/pion/webrtc/v2"
"github.com/pion/webrtc/v2/pkg/media"
)
@@ -60,7 +56,13 @@ var ssrcLock sync.Mutex
var playWaitList sync.Map
func init() {
m.RegisterCodec(NewRTPH264Codec(DefaultPayloadTypeH264, 90000))
m.RegisterCodec(NewRTPCodec(RTPCodecTypeVideo,
H264,
90000,
0,
"level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f",
DefaultPayloadTypeH264,
new(avformat.H264)))
//m.RegisterCodec(NewRTPPCMUCodec(DefaultPayloadTypePCMU, 8000))
api = NewAPI(WithMediaEngine(m))
InstallPlugin(&PluginConfig{
@@ -76,26 +78,8 @@ 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,
})
// codecs.H264Packet
// *os.File
}
func (rtc *WebRTC) Play(streamPath string) bool {
@@ -108,138 +92,43 @@ func (rtc *WebRTC) Play(streamPath string) bool {
}
case ICEConnectionStateConnected:
var sub Subscriber
var sps []byte
var pps []byte
sub.ID = rtc.RemoteAddr
sub.Type = "WebRTC"
var lastTimeStamp uint32
var dataBuilder bytes.Buffer
sub.OnData = func(packet *avformat.SendPacket) error {
if packet.Type == avformat.FLV_TAG_TYPE_AUDIO {
return nil
}
if packet.IsSequence {
payload := packet.Payload[11:]
spsLen := int(payload[0])<<8 + int(payload[1])
payload = payload[2:]
sps = payload[:spsLen]
payload = payload[1+spsLen:]
ppsLen := int(payload[0])<<8 + int(payload[1])
payload = payload[2:]
pps = payload[:ppsLen]
} else {
var s uint32
if lastTimeStamp > 0 {
s = packet.Timestamp - lastTimeStamp
}
if packet.IsKeyFrame {
dataBuilder.Write(avformat.NALU_Delimiter2)
dataBuilder.Write(sps)
dataBuilder.Write(avformat.NALU_Delimiter2)
dataBuilder.Write(pps)
rtc.videoTrack.WriteSample(media.Sample{
Data: sub.SPS,
Samples: 0,
})
rtc.videoTrack.WriteSample(media.Sample{
Data: sub.PPS,
Samples: 0,
})
}
payload := packet.Payload[5:]
for {
for payload := packet.Payload[5:]; len(payload) > 4; {
var naulLen = int(util.BigEndian.Uint32(payload))
payload = payload[4:]
dataBuilder.Write(avformat.NALU_Delimiter2)
dataBuilder.Write(payload[:naulLen])
rtc.videoTrack.WriteSample(media.Sample{
Data: dataBuilder.Bytes(),
Data: payload[:naulLen],
Samples: s * 90,
})
dataBuilder.Reset()
if len(payload) < naulLen+4 {
break
}
s = 0
payload = payload[naulLen:]
}
}
lastTimeStamp = packet.Timestamp
return nil
}
// stapA := func(naul ...[]byte) []byte {
// var buffer bytes.Buffer
// buffer.WriteByte((naul[0][0] & 224) | 24)
// for _, n := range naul {
// l := len(n)
// buffer.WriteByte(byte(l >> 8))
// buffer.WriteByte(byte(l))
// buffer.Write(n)
// }
// return buffer.Bytes()
// }
// //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
// }
// if packet.IsSequence {
// payload := packet.Payload[11:]
// spsLen := int(payload[0])<<8 + int(payload[1])
// sps = payload[2:spsLen]
// payload = payload[3+spsLen:]
// ppsLen := int(payload[0])<<8 + int(payload[1])
// pps = payload[2:ppsLen]
// } else {
// if packet.IsKeyFrame {
// 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
// }
// }
// payload := packet.Payload[5:]
// for {
// var naulLen = int(util.BigEndian.Uint32(payload))
// payload = payload[4:]
// _payload := payload[:naulLen]
// if naulLen > 1000 {
// indicator := (_payload[0] & 224) | 28
// nalutype := _payload[0] & 31
// header := 128 | nalutype
// part := _payload[1:1000]
// marker := false
// for {
// if err := rtc.WriteVideo(packet.Timestamp*90, marker, append([]byte{indicator, header}, part...)); err != nil {
// return err
// }
// if _payload == nil {
// break
// }
// _payload = _payload[1000:]
// if len(_payload) <= 1000 {
// header = 64 | nalutype
// part = _payload
// _payload = nil
// marker = true
// } else {
// header = nalutype
// part = _payload[:1000]
// }
// }
// } else {
// if err := rtc.WriteVideo(packet.Timestamp*90, true, _payload); err != nil {
// return err
// }
// }
// if len(payload) < naulLen+4 {
// break
// }
// payload = payload[naulLen:]
// }
// // if err := videoTrack.WriteRTP(&rtp.Packet{
// // Header: nextHeader(packet.Timestamp * 90),
// // Payload: aud,
// // }); err != nil {
// // return err
// // }
// }
// return nil
// }
go sub.Subscribe(streamPath)
}
})
@@ -299,7 +188,7 @@ func (rtc *WebRTC) Publish(streamPath string) bool {
if err = pack.Unmarshal(b[:i]); err != nil {
return
}
rtc.Unmarshal(pack.Payload)
// rtc.Unmarshal(pack.Payload)
// f.Write(bytes)
}
})