mirror of
https://github.com/Monibuca/plugin-webrtc.git
synced 2025-10-27 01:00:26 +08:00
开发播放功能
This commit is contained in:
langhuihui
260
main.go
260
main.go
@@ -111,118 +111,123 @@ func (rtc *WebRTC) Play(streamPath string) bool {
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Println(err)
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return false
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}
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peerConnection.OnICEConnectionStateChange(func(connectionState ICEConnectionState) {
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Printf("%s Connection State has changed %s ", streamPath, connectionState.String())
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switch connectionState {
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case ICEConnectionStateDisconnected:
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if rtc.Stream != nil {
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rtc.Stream.Close()
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}
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case ICEConnectionStateConnected:
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var sequence uint16
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var sub Subscriber
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sub.ID = rtc.RemoteAddr
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sub.Type = "WebRTC"
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nextHeader := func(ts uint32, marker bool) rtp.Header {
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sequence++
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return rtp.Header{
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Version: 2,
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SSRC: ssrc,
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PayloadType: DefaultPayloadTypeH264,
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SequenceNumber: sequence,
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Timestamp: ts,
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Marker: marker,
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}
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}
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stapA := func(naul ...[]byte) []byte {
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var buffer bytes.Buffer
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buffer.WriteByte(24)
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for _, n := range naul {
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l := len(n)
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buffer.WriteByte(byte(l >> 8))
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buffer.WriteByte(byte(l))
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buffer.Write(n)
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}
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return buffer.Bytes()
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}
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var sequence uint16
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var sub Subscriber
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var sps []byte
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var pps []byte
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sub.ID = rtc.RemoteAddr
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sub.Type = "WebRTC"
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nextHeader := func(ts uint32, marker bool) rtp.Header {
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sequence++
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return rtp.Header{
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Version: 2,
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SSRC: ssrc,
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PayloadType: DefaultPayloadTypeH264,
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SequenceNumber: sequence,
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Timestamp: ts,
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Marker: marker,
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}
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}
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stapA := func(naul ...[]byte) []byte {
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var buffer bytes.Buffer
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buffer.WriteByte(24)
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for _, n := range naul {
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l := len(n)
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buffer.WriteByte(byte(l >> 8))
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buffer.WriteByte(byte(l))
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buffer.Write(n)
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}
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return buffer.Bytes()
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}
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// aud := []byte{0x09, 0x30}
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sub.OnData = func(packet *avformat.SendPacket) error {
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if packet.Type == avformat.FLV_TAG_TYPE_AUDIO {
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return nil
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// aud := []byte{0x09, 0x30}
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sub.OnData = func(packet *avformat.SendPacket) error {
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if packet.Type == avformat.FLV_TAG_TYPE_AUDIO {
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return nil
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}
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if packet.IsSequence {
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payload := packet.Payload[11:]
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spsLen := int(payload[0])<<8 + int(payload[1])
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sps = payload[2:spsLen]
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payload = payload[3+spsLen:]
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ppsLen := int(payload[0])<<8 + int(payload[1])
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pps = payload[2:ppsLen]
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} else {
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if packet.IsKeyFrame {
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if err := videoTrack.WriteRTP(&rtp.Packet{
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Header: nextHeader(0, true),
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Payload: stapA(sps, pps),
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}); err != nil {
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return err
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}
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if packet.IsSequence {
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payload := packet.Payload[11:]
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spsLen := int(payload[0])<<8 + int(payload[1])
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sps := payload[2:spsLen]
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payload = payload[3+spsLen:]
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ppsLen := int(payload[0])<<8 + int(payload[1])
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pps := payload[2:ppsLen]
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}
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payload := packet.Payload[5:]
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for {
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var naulLen = int(util.BigEndian.Uint32(payload))
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payload = payload[4:]
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_payload := payload[:naulLen]
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if naulLen > 1000 {
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part := _payload[:1000]
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indicator := ((part[0] >> 5) << 5) | 28
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nalutype := part[0] & 31
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header := 128 | nalutype
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part = part[1:]
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marker := false
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for {
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if err := videoTrack.WriteRTP(&rtp.Packet{
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Header: nextHeader(packet.Timestamp*90, marker),
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Payload: append([]byte{indicator, header}, part...),
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}); err != nil {
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return err
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}
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if _payload == nil {
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break
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}
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if len(_payload[1000:]) <= 1000 {
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header = 64 | nalutype
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part = _payload[1000:]
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_payload = nil
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marker = true
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} else {
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header = nalutype
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part = _payload[1000:]
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_payload = part
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}
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}
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} else {
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if err := videoTrack.WriteRTP(&rtp.Packet{
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Header: nextHeader(0, false),
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Payload: stapA(sps, pps),
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Header: nextHeader(packet.Timestamp*90, true),
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Payload: _payload,
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}); err != nil {
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return err
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}
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} else {
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payload := packet.Payload[5:]
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for {
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var naulLen = int(util.BigEndian.Uint32(payload))
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payload = payload[4:]
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_payload := payload[:naulLen]
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if naulLen > 1000 {
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part := _payload[:1000]
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indicator := ((part[0] >> 5) << 5) | 28
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nalutype := part[0] & 31
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header := 128 | nalutype
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part = part[1:]
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marker := false
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for {
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if err := videoTrack.WriteRTP(&rtp.Packet{
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Header: nextHeader(packet.Timestamp*90, marker),
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Payload: append([]byte{indicator, header}, part...),
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}); err != nil {
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return err
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}
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marker = true
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if _payload == nil {
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break
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}
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if len(_payload[1000:]) <= 1000 {
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header = 64 | nalutype
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part = _payload[1000:]
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_payload = nil
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} else {
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header = nalutype
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part = _payload[1000:]
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_payload = part
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}
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}
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} else {
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if err := videoTrack.WriteRTP(&rtp.Packet{
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Header: nextHeader(packet.Timestamp*90, false),
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Payload: _payload,
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}); err != nil {
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return err
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}
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}
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if len(payload) < naulLen+4 {
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break
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}
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payload = payload[naulLen:]
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}
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// if err := videoTrack.WriteRTP(&rtp.Packet{
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// Header: nextHeader(packet.Timestamp * 90),
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// Payload: aud,
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// }); err != nil {
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// return err
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// }
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}
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return nil
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if len(payload) < naulLen+4 {
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break
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}
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payload = payload[naulLen:]
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}
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sub.Subscribe(streamPath)
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// if err := videoTrack.WriteRTP(&rtp.Packet{
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// Header: nextHeader(packet.Timestamp * 90),
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// Payload: aud,
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// }); err != nil {
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// return err
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// }
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}
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})
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return nil
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}
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go sub.Subscribe(streamPath)
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// peerConnection.OnICEConnectionStateChange(func(connectionState ICEConnectionState) {
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// Printf("%s Connection State has changed %s ", streamPath, connectionState.String())
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// switch connectionState {
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// case ICEConnectionStateDisconnected:
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// if rtc.Stream != nil {
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// rtc.Stream.Close()
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// }
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// case ICEConnectionStateConnected:
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// }
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// })
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return true
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}
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func (rtc *WebRTC) Publish(streamPath string) bool {
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@@ -289,24 +294,13 @@ func (rtc *WebRTC) Publish(streamPath string) bool {
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}
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return true
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}
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func (rtc *WebRTC) GetAnswer(offer SessionDescription) ([]byte, error) {
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if err := rtc.SetRemoteDescription(offer); err != nil {
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Println(err)
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return nil, err
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}
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// Create answer
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answer, err := rtc.CreateAnswer(nil)
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if err != nil {
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Println(err)
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return nil, err
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}
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func (rtc *WebRTC) GetAnswer(localSdp SessionDescription) ([]byte, error) {
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// Sets the LocalDescription, and starts our UDP listeners
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if err = rtc.SetLocalDescription(answer); err != nil {
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if err := rtc.SetLocalDescription(localSdp); err != nil {
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Println(err)
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return nil, err
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}
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if bytes, err := json.Marshal(answer); err != nil {
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if bytes, err := json.Marshal(localSdp); err != nil {
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Println(err)
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return bytes, err
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} else {
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@@ -316,17 +310,22 @@ func (rtc *WebRTC) GetAnswer(offer SessionDescription) ([]byte, error) {
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func run() {
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http.HandleFunc("/webrtc/play", func(w http.ResponseWriter, r *http.Request) {
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streamPath := r.URL.Query().Get("streamPath")
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offer := SessionDescription{}
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bytes, err := ioutil.ReadAll(r.Body)
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err = json.Unmarshal(bytes, &offer)
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if err != nil {
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Println(err)
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return
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}
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// offer := SessionDescription{}
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// bytes, err := ioutil.ReadAll(r.Body)
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// err = json.Unmarshal(bytes, &offer)
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// if err != nil {
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// Println(err)
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// return
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// }
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rtc := new(WebRTC)
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rtc.RemoteAddr = r.RemoteAddr
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if rtc.Play(streamPath) {
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if bytes, err = rtc.GetAnswer(offer); err == nil {
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offer, err := rtc.CreateOffer(nil)
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if err != nil {
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Println(err)
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return
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}
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if bytes, err := rtc.GetAnswer(offer); err == nil {
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w.Write(bytes)
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} else {
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Println(err)
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@@ -349,7 +348,16 @@ func run() {
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rtc := new(WebRTC)
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rtc.RemoteAddr = r.RemoteAddr
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if rtc.Publish(streamPath) {
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if bytes, err = rtc.GetAnswer(offer); err == nil {
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if err := rtc.SetRemoteDescription(offer); err != nil {
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Println(err)
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return
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}
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answer, err := rtc.CreateAnswer(nil)
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if err != nil {
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Println(err)
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return
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}
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if bytes, err = rtc.GetAnswer(answer); err == nil {
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w.Write(bytes)
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} else {
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Println(err)
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