mirror of
https://github.com/AlexxIT/go2rtc.git
synced 2025-10-05 16:26:50 +08:00
252 lines
5.0 KiB
Go
252 lines
5.0 KiB
Go
package homekit
|
|
|
|
import (
|
|
"errors"
|
|
"fmt"
|
|
"math/rand"
|
|
"net"
|
|
"net/url"
|
|
"time"
|
|
|
|
"github.com/AlexxIT/go2rtc/pkg/core"
|
|
"github.com/AlexxIT/go2rtc/pkg/hap"
|
|
"github.com/AlexxIT/go2rtc/pkg/hap/camera"
|
|
"github.com/AlexxIT/go2rtc/pkg/srtp"
|
|
"github.com/pion/rtp"
|
|
)
|
|
|
|
type Client struct {
|
|
core.SuperProducer
|
|
|
|
hap *hap.Client
|
|
srtp *srtp.Server
|
|
|
|
videoConfig camera.SupportedVideoStreamConfig
|
|
audioConfig camera.SupportedAudioStreamConfig
|
|
|
|
videoSession *srtp.Session
|
|
audioSession *srtp.Session
|
|
|
|
stream *camera.Stream
|
|
}
|
|
|
|
func Dial(rawURL string, server *srtp.Server) (*Client, error) {
|
|
u, err := url.Parse(rawURL)
|
|
if err != nil {
|
|
return nil, err
|
|
}
|
|
|
|
query := u.Query()
|
|
conn := &hap.Client{
|
|
DeviceAddress: u.Host,
|
|
DeviceID: query.Get("device_id"),
|
|
DevicePublic: hap.DecodeKey(query.Get("device_public")),
|
|
ClientID: query.Get("client_id"),
|
|
ClientPrivate: hap.DecodeKey(query.Get("client_private")),
|
|
}
|
|
|
|
if err = conn.Dial(); err != nil {
|
|
return nil, err
|
|
}
|
|
|
|
client := &Client{
|
|
SuperProducer: core.SuperProducer{
|
|
Type: "HomeKit active producer",
|
|
URL: conn.URL(),
|
|
},
|
|
hap: conn,
|
|
srtp: server,
|
|
}
|
|
|
|
return client, nil
|
|
}
|
|
|
|
func (c *Client) Conn() net.Conn {
|
|
return c.hap.Conn
|
|
}
|
|
|
|
func (c *Client) GetMedias() []*core.Media {
|
|
if c.Medias != nil {
|
|
return c.Medias
|
|
}
|
|
|
|
acc, err := c.hap.GetFirstAccessory()
|
|
if err != nil {
|
|
return nil
|
|
}
|
|
|
|
char := acc.GetCharacter(camera.TypeSupportedVideoStreamConfiguration)
|
|
if char == nil {
|
|
return nil
|
|
}
|
|
if err = char.ReadTLV8(&c.videoConfig); err != nil {
|
|
return nil
|
|
}
|
|
|
|
char = acc.GetCharacter(camera.TypeSupportedAudioStreamConfiguration)
|
|
if char == nil {
|
|
return nil
|
|
}
|
|
if err = char.ReadTLV8(&c.audioConfig); err != nil {
|
|
return nil
|
|
}
|
|
|
|
c.URL = c.hap.URL()
|
|
c.SDP = fmt.Sprintf("%+v\n%+v", c.videoConfig, c.audioConfig)
|
|
|
|
c.Medias = []*core.Media{
|
|
videoToMedia(c.videoConfig.Codecs),
|
|
audioToMedia(c.audioConfig.Codecs),
|
|
{
|
|
Kind: core.KindVideo,
|
|
Direction: core.DirectionRecvonly,
|
|
Codecs: []*core.Codec{
|
|
{
|
|
Name: core.CodecJPEG,
|
|
ClockRate: 90000,
|
|
PayloadType: core.PayloadTypeRAW,
|
|
},
|
|
},
|
|
},
|
|
}
|
|
|
|
return c.Medias
|
|
}
|
|
|
|
func (c *Client) Start() error {
|
|
if c.Receivers == nil {
|
|
return errors.New("producer without tracks")
|
|
}
|
|
|
|
if c.Receivers[0].Codec.Name == core.CodecJPEG {
|
|
return c.startMJPEG()
|
|
}
|
|
|
|
videoTrack := c.trackByKind(core.KindVideo)
|
|
videoCodec := trackToVideo(videoTrack, &c.videoConfig.Codecs[0])
|
|
|
|
audioTrack := c.trackByKind(core.KindAudio)
|
|
audioCodec := trackToAudio(audioTrack, &c.audioConfig.Codecs[0])
|
|
|
|
c.videoSession = &srtp.Session{Local: c.srtpEndpoint()}
|
|
c.audioSession = &srtp.Session{Local: c.srtpEndpoint()}
|
|
|
|
var err error
|
|
c.stream, err = camera.NewStream(c.hap, videoCodec, audioCodec, c.videoSession, c.audioSession)
|
|
if err != nil {
|
|
return err
|
|
}
|
|
|
|
c.srtp.AddSession(c.videoSession)
|
|
c.srtp.AddSession(c.audioSession)
|
|
|
|
deadline := time.NewTimer(core.ConnDeadline)
|
|
|
|
if videoTrack != nil {
|
|
c.videoSession.OnReadRTP = func(packet *rtp.Packet) {
|
|
deadline.Reset(core.ConnDeadline)
|
|
videoTrack.WriteRTP(packet)
|
|
c.Recv += len(packet.Payload)
|
|
}
|
|
|
|
if audioTrack != nil {
|
|
c.audioSession.OnReadRTP = func(packet *rtp.Packet) {
|
|
audioTrack.WriteRTP(packet)
|
|
c.Recv += len(packet.Payload)
|
|
}
|
|
}
|
|
} else {
|
|
c.audioSession.OnReadRTP = func(packet *rtp.Packet) {
|
|
deadline.Reset(core.ConnDeadline)
|
|
audioTrack.WriteRTP(packet)
|
|
c.Recv += len(packet.Payload)
|
|
}
|
|
}
|
|
|
|
if c.audioSession.OnReadRTP != nil {
|
|
c.audioSession.OnReadRTP = timekeeper(c.audioSession.OnReadRTP)
|
|
}
|
|
|
|
<-deadline.C
|
|
|
|
return nil
|
|
}
|
|
|
|
func (c *Client) Stop() error {
|
|
_ = c.SuperProducer.Close()
|
|
|
|
if c.videoSession != nil && c.videoSession.Remote != nil {
|
|
c.srtp.DelSession(c.videoSession)
|
|
}
|
|
if c.audioSession != nil && c.audioSession.Remote != nil {
|
|
c.srtp.DelSession(c.audioSession)
|
|
}
|
|
|
|
return c.hap.Close()
|
|
}
|
|
|
|
func (c *Client) trackByKind(kind string) *core.Receiver {
|
|
for _, receiver := range c.Receivers {
|
|
if receiver.Codec.Kind() == kind {
|
|
return receiver
|
|
}
|
|
}
|
|
return nil
|
|
}
|
|
|
|
func (c *Client) startMJPEG() error {
|
|
receiver := c.Receivers[0]
|
|
|
|
for {
|
|
b, err := c.hap.GetImage(1920, 1080)
|
|
if err != nil {
|
|
return err
|
|
}
|
|
|
|
c.Recv += len(b)
|
|
|
|
packet := &rtp.Packet{
|
|
Header: rtp.Header{Timestamp: core.Now90000()},
|
|
Payload: b,
|
|
}
|
|
receiver.WriteRTP(packet)
|
|
}
|
|
}
|
|
|
|
func (c *Client) srtpEndpoint() *srtp.Endpoint {
|
|
return &srtp.Endpoint{
|
|
Addr: c.hap.LocalIP(),
|
|
Port: uint16(c.srtp.Port()),
|
|
MasterKey: []byte(core.RandString(16, 0)),
|
|
MasterSalt: []byte(core.RandString(14, 0)),
|
|
SSRC: rand.Uint32(),
|
|
}
|
|
}
|
|
|
|
func timekeeper(handler core.HandlerFunc) core.HandlerFunc {
|
|
const sampleRate = 16000
|
|
const sampleSize = 480
|
|
|
|
var send time.Duration
|
|
var firstTime time.Time
|
|
|
|
return func(packet *rtp.Packet) {
|
|
now := time.Now()
|
|
|
|
if send != 0 {
|
|
elapsed := now.Sub(firstTime) * sampleRate / time.Second
|
|
if send+sampleSize > elapsed {
|
|
return // drop overflow frame
|
|
}
|
|
} else {
|
|
firstTime = now
|
|
}
|
|
|
|
send += sampleSize
|
|
|
|
packet.Timestamp = uint32(send)
|
|
|
|
handler(packet)
|
|
}
|
|
}
|