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https://github.com/m1k1o/neko.git
synced 2024-07-24 14:40:50 +12:00
Implement Audio & Video using custom StreamManager.
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@ -19,48 +19,48 @@ import (
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func New(desktop types.DesktopManager, capture types.CaptureManager, config *config.WebRTC) *WebRTCManagerCtx {
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return &WebRTCManagerCtx{
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logger: log.With().Str("module", "webrtc").Logger(),
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desktop: desktop,
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capture: capture,
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config: config,
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logger: log.With().Str("module", "webrtc").Logger(),
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videoCodec: capture.Video().Codec(),
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audioCodec: capture.Audio().Codec(),
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desktop: desktop,
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capture: capture,
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config: config,
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}
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}
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type WebRTCManagerCtx struct {
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logger zerolog.Logger
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videoTrack *webrtc.TrackLocalStaticSample
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audioTrack *webrtc.TrackLocalStaticSample
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videoCodec codec.RTPCodec
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audioCodec codec.RTPCodec
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desktop types.DesktopManager
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capture types.CaptureManager
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config *config.WebRTC
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logger zerolog.Logger
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videoTrack *webrtc.TrackLocalStaticSample
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audioTrack *webrtc.TrackLocalStaticSample
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videoCodec codec.RTPCodec
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audioCodec codec.RTPCodec
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desktop types.DesktopManager
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capture types.CaptureManager
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config *config.WebRTC
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}
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func (manager *WebRTCManagerCtx) Start() {
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var err error
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// create audio track
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manager.audioCodec = manager.capture.AudioCodec()
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manager.audioTrack, err = webrtc.NewTrackLocalStaticSample(manager.audioCodec.Capability, "audio", "stream")
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if err != nil {
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manager.logger.Panic().Err(err).Msg("unable to create audio track")
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}
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manager.capture.OnAudioFrame(func(sample types.Sample) {
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manager.capture.Audio().OnSample(func(sample types.Sample) {
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if err := manager.audioTrack.WriteSample(media.Sample(sample)); err != nil && err != io.ErrClosedPipe {
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manager.logger.Warn().Err(err).Msg("audio pipeline failed to write")
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}
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})
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// create video track
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manager.videoCodec = manager.capture.VideoCodec()
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manager.videoTrack, err = webrtc.NewTrackLocalStaticSample(manager.videoCodec.Capability, "video", "stream")
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if err != nil {
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manager.logger.Panic().Err(err).Msg("unable to create video track")
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}
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manager.capture.OnVideoFrame(func(sample types.Sample) {
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manager.capture.Video().OnSample(func(sample types.Sample) {
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if err := manager.videoTrack.WriteSample(media.Sample(sample)); err != nil && err != io.ErrClosedPipe {
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manager.logger.Warn().Err(err).Msg("video pipeline failed to write")
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}
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@ -91,6 +91,7 @@ func (manager *WebRTCManagerCtx) ICEServers() []string {
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func (manager *WebRTCManagerCtx) CreatePeer(session types.Session) (*webrtc.SessionDescription, error) {
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logger := manager.logger.With().Str("id", session.ID()).Logger()
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// Create MediaEngine
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engine, err := manager.mediaEngine()
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if err != nil {
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return nil, err
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@ -129,7 +130,21 @@ func (manager *WebRTCManagerCtx) CreatePeer(session types.Session) (*webrtc.Sess
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})
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}
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if err := manager.registerTracks(connection); err != nil {
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audioTransceiver, err := connection.AddTransceiverFromTrack(manager.audioTrack, webrtc.RtpTransceiverInit{
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Direction: webrtc.RTPTransceiverDirectionSendonly,
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})
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if err != nil {
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return nil, err
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}
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videoTransceiver, err := connection.AddTransceiverFromTrack(manager.videoTrack, webrtc.RtpTransceiverInit{
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Direction: webrtc.RTPTransceiverDirectionSendonly,
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})
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if err != nil {
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return nil, err
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}
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if _, err := connection.CreateDataChannel("data", nil); err != nil {
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return nil, err
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}
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@ -179,18 +194,19 @@ func (manager *WebRTCManagerCtx) CreatePeer(session types.Session) (*webrtc.Sess
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})
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session.SetWebRTCPeer(&WebRTCPeerCtx{
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api: api,
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engine: engine,
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settings: settings,
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connection: connection,
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configuration: configuration,
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api: api,
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engine: engine,
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settings: settings,
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connection: connection,
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configuration: configuration,
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audioTransceiver: audioTransceiver,
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videoTransceiver: videoTransceiver,
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})
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return connection.LocalDescription(), nil
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}
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func (manager *WebRTCManagerCtx) mediaEngine() (*webrtc.MediaEngine, error) {
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// Create MediaEngine
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engine := &webrtc.MediaEngine{}
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if err := manager.videoCodec.Register(engine); err != nil {
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@ -235,20 +251,3 @@ func (manager *WebRTCManagerCtx) apiConfiguration() *webrtc.Configuration {
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SDPSemantics: webrtc.SDPSemanticsUnifiedPlanWithFallback,
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}
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}
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func (manager *WebRTCManagerCtx) registerTracks(connection *webrtc.PeerConnection) error {
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if _, err := connection.AddTransceiverFromTrack(manager.videoTrack, webrtc.RtpTransceiverInit{
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Direction: webrtc.RTPTransceiverDirectionSendonly,
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}); err != nil {
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return err
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}
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if _, err := connection.AddTransceiverFromTrack(manager.audioTrack, webrtc.RtpTransceiverInit{
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Direction: webrtc.RTPTransceiverDirectionSendonly,
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}); err != nil {
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return err
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}
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_, err := connection.CreateDataChannel("data", nil)
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return err
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}
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@ -3,11 +3,13 @@ package webrtc
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import "github.com/pion/webrtc/v3"
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type WebRTCPeerCtx struct {
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api *webrtc.API
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engine *webrtc.MediaEngine
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settings *webrtc.SettingEngine
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connection *webrtc.PeerConnection
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configuration *webrtc.Configuration
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api *webrtc.API
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engine *webrtc.MediaEngine
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settings *webrtc.SettingEngine
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connection *webrtc.PeerConnection
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configuration *webrtc.Configuration
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audioTransceiver *webrtc.RTPTransceiver
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videoTransceiver *webrtc.RTPTransceiver
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}
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func (webrtc_peer *WebRTCPeerCtx) SignalAnswer(sdp string) error {
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@ -21,6 +23,14 @@ func (webrtc_peer *WebRTCPeerCtx) SignalCandidate(candidate webrtc.ICECandidateI
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return webrtc_peer.connection.AddICECandidate(candidate)
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}
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func (webrtc_peer *WebRTCPeerCtx) ReplaceAudioTrack(track webrtc.TrackLocal) error {
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return webrtc_peer.audioTransceiver.Sender().ReplaceTrack(track)
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}
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func (webrtc_peer *WebRTCPeerCtx) ReplaceVideoTrack(track webrtc.TrackLocal) error {
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return webrtc_peer.videoTransceiver.Sender().ReplaceTrack(track)
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}
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func (webrtc_peer *WebRTCPeerCtx) Destroy() error {
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if webrtc_peer.connection == nil || webrtc_peer.connection.ConnectionState() != webrtc.PeerConnectionStateConnected {
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return nil
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