public async Task CaptureMedia(long fromUid, bool r = false)
{
LocalMedia = Media.CreateMedia(); //创建一个Media对象
RTCMediaStreamConstraints mediaStreamConstraints = new RTCMediaStreamConstraints() //设置要获取的流
{
audioEnabled = true,
videoEnabled = true
};
//音频播放
var apd = LocalMedia.GetAudioPlayoutDevices();
if (apd.Count > 0)
{
LocalMedia.SelectAudioPlayoutDevice(apd[0]);
}
if (fromUid == 0)
{
//音频捕获
var acd = LocalMedia.GetAudioCaptureDevices();
if (acd.Count > 0)
{
mediaStreamConstraints.audioEnabled = true;
LocalMedia.SelectAudioCaptureDevice(acd[0]);
}
//视频捕获
var vcd = LocalMedia.GetVideoCaptureDevices();
if (vcd.Count > 0)
{
mediaStreamConstraints.videoEnabled = true;
LocalMedia.SelectVideoDevice(vcd.First(p => p.Location.Panel == Windows.Devices.Enumeration.Panel.Front));//设置视频捕获设备
}
}
var mediaStream = await LocalMedia.GetUserMedia(mediaStreamConstraints);//获取视频流 这里视频和音频是一起传输的
if (fromUid == 0)
{
var videotracs = mediaStream.GetVideoTracks();
if (videotracs.Count > 0)
{
var source = LocalMedia.CreateMediaSource(videotracs.FirstOrDefault(), mediaStream.Id); //创建播放源
LocalMediaPlayer.SetMediaStreamSource(source); //设置MediaElement的播放源
LocalMediaPlayer.Play();
}
await CreatePublisher(mediaStream);
//await CreateServer(mediaStream);
}
else
{
await CreateReceiver(mediaStream, fromUid);
}
}
/// <summary>
/// Creates a peer connection.
/// </summary>
/// <returns>True if connection to a peer is successfully created.</returns>
private async Task <bool> CreatePeerConnection(CancellationToken cancelationToken)
{
Debug.Assert(_peerConnection == null);
if (cancelationToken.IsCancellationRequested)
{
return(false);
}
var config = new RTCConfiguration()
{
BundlePolicy = RTCBundlePolicy.Balanced,
#if ORTCLIB
SignalingMode = _signalingMode,
GatherOptions = new RTCIceGatherOptions()
{
IceServers = new List <RTCIceServer>(_iceServers),
}
#else
IceTransportPolicy = RTCIceTransportPolicy.All,
IceServers = _iceServers
#endif
};
Debug.WriteLine("Conductor: Creating peer connection.");
_peerConnection = new RTCPeerConnection(config);
if (_peerConnection == null)
{
throw new NullReferenceException("Peer connection is not created.");
}
#if !ORTCLIB
_peerConnection.EtwStatsEnabled = _etwStatsEnabled;
_peerConnection.ConnectionHealthStatsEnabled = _peerConnectionStatsEnabled;
#endif
if (cancelationToken.IsCancellationRequested)
{
return(false);
}
#if ORTCLIB
OrtcStatsManager.Instance.Initialize(_peerConnection);
#endif
OnPeerConnectionCreated?.Invoke();
_peerConnection.OnIceCandidate += PeerConnection_OnIceCandidate;
#if ORTCLIB
_peerConnection.OnTrack += PeerConnection_OnAddTrack;
_peerConnection.OnTrackGone += PeerConnection_OnRemoveTrack;
_peerConnection.OnIceConnectionStateChange += () => { Debug.WriteLine("Conductor: Ice connection state change, state=" + (null != _peerConnection ? _peerConnection.IceConnectionState.ToString() : "closed")); };
#else
_peerConnection.OnAddStream += PeerConnection_OnAddStream;
_peerConnection.OnRemoveStream += PeerConnection_OnRemoveStream;
_peerConnection.OnConnectionHealthStats += PeerConnection_OnConnectionHealthStats;
#endif
Debug.WriteLine("Conductor: Getting user media.");
RTCMediaStreamConstraints mediaStreamConstraints = new RTCMediaStreamConstraints
{
// Always include audio/video enabled in the media stream,
// so it will be possible to enable/disable audio/video if
// the call was initiated without microphone/camera
audioEnabled = true,
videoEnabled = true
};
if (cancelationToken.IsCancellationRequested)
{
return(false);
}
#if ORTCLIB
var tracks = await _media.GetUserMedia(mediaStreamConstraints);
if (tracks != null)
{
RTCRtpCapabilities audioCapabilities = RTCRtpSender.GetCapabilities("audio");
RTCRtpCapabilities videoCapabilities = RTCRtpSender.GetCapabilities("video");
_mediaStream = new MediaStream(tracks);
Debug.WriteLine("Conductor: Adding local media stream.");
IList <MediaStream> mediaStreamList = new List <MediaStream>();
mediaStreamList.Add(_mediaStream);
foreach (var mediaStreamTrack in tracks)
{
//Create stream track configuration based on capabilities
RTCMediaStreamTrackConfiguration configuration = null;
if (mediaStreamTrack.Kind == MediaStreamTrackKind.Audio && audioCapabilities != null)
{
configuration =
await Helper.GetTrackConfigurationForCapabilities(audioCapabilities, AudioCodec);
}
else if (mediaStreamTrack.Kind == MediaStreamTrackKind.Video && videoCapabilities != null)
{
configuration =
await Helper.GetTrackConfigurationForCapabilities(videoCapabilities, VideoCodec);
}
if (configuration != null)
{
_peerConnection.AddTrack(mediaStreamTrack, mediaStreamList, configuration);
}
}
}
#else
_mediaStream = await _media.GetUserMedia(mediaStreamConstraints);
#endif
if (cancelationToken.IsCancellationRequested)
{
return(false);
}
#if !ORTCLIB
Debug.WriteLine("Conductor: Adding local media stream.");
_peerConnection.AddStream(_mediaStream);
#endif
OnAddLocalStream?.Invoke(new MediaStreamEvent()
{
Stream = _mediaStream
});
if (cancelationToken.IsCancellationRequested)
{
return(false);
}
return(true);
}
public SymplePlayerEngineWebRTC(SymplePlayer player) : base(player)
{
Messenger.Broadcast(SympleLog.LogDebug, "symple:webrtc: init");
#if NETFX_CORE
if (!webrtcInitialized)
{
// needed before calling any webrtc functions http://stackoverflow.com/questions/43331677/webrtc-for-uwp-new-rtcpeerconnection-doesnt-complete-execution
if (player.options.CoreDispatcher != null)
{
WebRTC.Initialize(player.options.CoreDispatcher);
}
else
{
WebRTC.Initialize(null);
}
WebRTC.EnableLogging(LogLevel.LOGLVL_ERROR);
WebRTC.EnableLogging(LogLevel.LOGLVL_INFO);
WebRTC.EnableLogging(LogLevel.LOGLVL_SENSITIVE);
WebRTC.EnableLogging(LogLevel.LOGLVL_VERBOSE);
WebRTC.EnableLogging(LogLevel.LOGLVL_WARNING);
Messenger.Broadcast(SympleLog.LogInfo, "WebRTC logging enabled, log folder = " + WebRTC.LogFolder.Path + ", filename = " + WebRTC.LogFileName);
webrtcInitialized = true;
}
this.userMediaConstraints = player.options.userMediaConstraints;
if (player.options.rtcConfig != null)
{
this.rtcConfig = player.options.rtcConfig;
}
else
{
this.rtcConfig = new RTCConfiguration();
this.rtcConfig.IceServers.Add(new RTCIceServer()
{
Url = "stun:stun.l.google.com:19302", Username = string.Empty, Credential = string.Empty
});
}
#endif
/*
* this.rtcOptions = player.options.rtcOptions || {
* optional: [
* { DtlsSrtpKeyAgreement: true } // required for FF <=> Chrome interop
* ]
* };
*/
// Specifies that this client will be the ICE initiator,
// and will be sending the initial SDP Offer.
this.initiator = player.options.initiator;
Messenger.Broadcast(SympleLog.LogDebug, "symple:webrtc: constructor, set this.initiator to " + this.initiator);
#if NETFX_CORE
// Reference to the active local or remote media stream
this.activeStream = null;
#endif
}
private async Task Initialize()
{
//Initialization of WebRTC worker threads, etc
Org.WebRtc.WebRTC.Initialize(Dispatcher);
_media = Media.CreateMedia();
//Selecting video device to use, setting preferred capabilities
var videoDevices = _media.GetVideoCaptureDevices();
var selectedVideoDevice = videoDevices.First();
var videoCapabilites = await selectedVideoDevice.GetVideoCaptureCapabilities();
var selectedVideoCapability = videoCapabilites.FirstOrDefault();
//Needed for HoloLens camera, will not set compatible video capability automatically
//Hololens Cam default capability: 1280x720x30
Org.WebRtc.WebRTC.SetPreferredVideoCaptureFormat(
(int)selectedVideoCapability.Width,
(int)selectedVideoCapability.Height,
(int)selectedVideoCapability.FrameRate);
//Setting up local stream
RTCMediaStreamConstraints mediaStreamConstraints = new RTCMediaStreamConstraints
{
audioEnabled = false,
videoEnabled = true
};
_localStream = await _media.GetUserMedia(mediaStreamConstraints);
_media.SelectVideoDevice(selectedVideoDevice);
// Get Video Tracks
var videotrac = _localStream.GetVideoTracks();
foreach (var videoTrack in videotrac) //This foreach may not be necessary
{
videoTrack.Enabled = true;
}
var selectedVideoTrac = videotrac.FirstOrDefault();
Debug.WriteLine("Creating RTCPeerConnection");
var config = new RTCConfiguration()
{
BundlePolicy = RTCBundlePolicy.Balanced,
IceTransportPolicy = RTCIceTransportPolicy.All,
IceServers = GetDefaultList()
};
_peerConnection = new RTCPeerConnection(config);
_peerConnection.OnIceCandidate += _localRtcPeerConnection_OnIceCandidate;
_peerConnection.OnIceConnectionChange += _localRtcPeerConnection_OnIceConnectionChange;
_peerConnection.OnAddStream += _peerConnection_OnAddStream;
//_peerConnection.AddStream(_localStream);
_media.AddVideoTrackMediaElementPair(selectedVideoTrac, _localVideo, _localStream.Id);
// Send event started
Element.SendStarted();
//Debug.WriteLine("Creating 'remote' RTCPeerConnection");
//_remoteRtcPeerConnection = new RTCPeerConnection(config);
//_remoteRtcPeerConnection.OnIceCandidate += _remoteRtcPeerConnection_OnIceCandidate;
//_remoteRtcPeerConnection.OnIceConnectionChange += _remoteRtcPeerConnection_OnIceConnectionChange;
//// Wait for Stream
//_remoteRtcPeerConnection.OnAddStream += _remoteRtcPeerConnection_OnAddStream;
}
