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1 /* | 1 /* |
2 * libjingle | 2 * libjingle |
3 * Copyright 2012 Google Inc. | 3 * Copyright 2012 Google Inc. |
4 * | 4 * |
5 * Redistribution and use in source and binary forms, with or without | 5 * Redistribution and use in source and binary forms, with or without |
6 * modification, are permitted provided that the following conditions are met: | 6 * modification, are permitted provided that the following conditions are met: |
7 * | 7 * |
8 * 1. Redistributions of source code must retain the above copyright notice, | 8 * 1. Redistributions of source code must retain the above copyright notice, |
9 * this list of conditions and the following disclaimer. | 9 * this list of conditions and the following disclaimer. |
10 * 2. Redistributions in binary form must reproduce the above copyright notice, | 10 * 2. Redistributions in binary form must reproduce the above copyright notice, |
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26 */ | 26 */ |
27 | 27 |
28 #include "talk/app/webrtc/peerconnection.h" | 28 #include "talk/app/webrtc/peerconnection.h" |
29 | 29 |
30 #include <vector> | 30 #include <vector> |
31 | 31 |
32 #include "talk/app/webrtc/dtmfsender.h" | 32 #include "talk/app/webrtc/dtmfsender.h" |
33 #include "talk/app/webrtc/jsepicecandidate.h" | 33 #include "talk/app/webrtc/jsepicecandidate.h" |
34 #include "talk/app/webrtc/jsepsessiondescription.h" | 34 #include "talk/app/webrtc/jsepsessiondescription.h" |
35 #include "talk/app/webrtc/mediaconstraintsinterface.h" | 35 #include "talk/app/webrtc/mediaconstraintsinterface.h" |
36 #include "talk/app/webrtc/mediastreamhandler.h" | 36 #include "talk/app/webrtc/rtpreceiver.h" |
37 #include "talk/app/webrtc/rtpsender.h" | |
37 #include "talk/app/webrtc/streamcollection.h" | 38 #include "talk/app/webrtc/streamcollection.h" |
38 #include "webrtc/p2p/client/basicportallocator.h" | 39 #include "webrtc/p2p/client/basicportallocator.h" |
39 #include "talk/session/media/channelmanager.h" | 40 #include "talk/session/media/channelmanager.h" |
40 #include "webrtc/base/logging.h" | 41 #include "webrtc/base/logging.h" |
41 #include "webrtc/base/stringencode.h" | 42 #include "webrtc/base/stringencode.h" |
42 #include "webrtc/system_wrappers/interface/field_trial.h" | 43 #include "webrtc/system_wrappers/interface/field_trial.h" |
43 | 44 |
44 namespace { | 45 namespace { |
45 | 46 |
46 using webrtc::PeerConnectionInterface; | 47 using webrtc::PeerConnectionInterface; |
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332 observer_(NULL), | 333 observer_(NULL), |
333 uma_observer_(NULL), | 334 uma_observer_(NULL), |
334 signaling_state_(kStable), | 335 signaling_state_(kStable), |
335 ice_state_(kIceNew), | 336 ice_state_(kIceNew), |
336 ice_connection_state_(kIceConnectionNew), | 337 ice_connection_state_(kIceConnectionNew), |
337 ice_gathering_state_(kIceGatheringNew) { | 338 ice_gathering_state_(kIceGatheringNew) { |
338 } | 339 } |
339 | 340 |
340 PeerConnection::~PeerConnection() { | 341 PeerConnection::~PeerConnection() { |
341 ASSERT(signaling_thread()->IsCurrent()); | 342 ASSERT(signaling_thread()->IsCurrent()); |
342 if (mediastream_signaling_) | 343 if (mediastream_signaling_) { |
343 mediastream_signaling_->TearDown(); | 344 mediastream_signaling_->TearDown(); |
344 if (stream_handler_container_) | 345 } |
345 stream_handler_container_->TearDown(); | 346 // Need to detach RTP senders/receivers from WebRtcSession, |
347 // since it's about to be destroyed. | |
348 for (const auto& sender : rtp_senders_) { | |
349 sender->Stop(); | |
350 } | |
351 for (const auto& receiver : rtp_receivers_) { | |
352 receiver->Stop(); | |
353 } | |
346 } | 354 } |
347 | 355 |
348 bool PeerConnection::Initialize( | 356 bool PeerConnection::Initialize( |
349 const PeerConnectionInterface::RTCConfiguration& configuration, | 357 const PeerConnectionInterface::RTCConfiguration& configuration, |
350 const MediaConstraintsInterface* constraints, | 358 const MediaConstraintsInterface* constraints, |
351 PortAllocatorFactoryInterface* allocator_factory, | 359 PortAllocatorFactoryInterface* allocator_factory, |
352 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store, | 360 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store, |
353 PeerConnectionObserver* observer) { | 361 PeerConnectionObserver* observer) { |
354 ASSERT(observer != NULL); | 362 ASSERT(observer != NULL); |
355 if (!observer) | 363 if (!observer) |
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392 port_allocator_->set_step_delay(cricket::kMinimumStepDelay); | 400 port_allocator_->set_step_delay(cricket::kMinimumStepDelay); |
393 | 401 |
394 mediastream_signaling_.reset(new MediaStreamSignaling( | 402 mediastream_signaling_.reset(new MediaStreamSignaling( |
395 factory_->signaling_thread(), this, factory_->channel_manager())); | 403 factory_->signaling_thread(), this, factory_->channel_manager())); |
396 | 404 |
397 session_.reset(new WebRtcSession(factory_->channel_manager(), | 405 session_.reset(new WebRtcSession(factory_->channel_manager(), |
398 factory_->signaling_thread(), | 406 factory_->signaling_thread(), |
399 factory_->worker_thread(), | 407 factory_->worker_thread(), |
400 port_allocator_.get(), | 408 port_allocator_.get(), |
401 mediastream_signaling_.get())); | 409 mediastream_signaling_.get())); |
402 stream_handler_container_.reset(new MediaStreamHandlerContainer( | |
403 session_.get(), session_.get())); | |
404 stats_.reset(new StatsCollector(session_.get())); | 410 stats_.reset(new StatsCollector(session_.get())); |
405 | 411 |
406 // Initialize the WebRtcSession. It creates transport channels etc. | 412 // Initialize the WebRtcSession. It creates transport channels etc. |
407 if (!session_->Initialize(factory_->options(), constraints, | 413 if (!session_->Initialize(factory_->options(), constraints, |
408 dtls_identity_store.Pass(), configuration)) | 414 dtls_identity_store.Pass(), configuration)) |
409 return false; | 415 return false; |
410 | 416 |
411 // Register PeerConnection as receiver of local ice candidates. | 417 // Register PeerConnection as receiver of local ice candidates. |
412 // All the callbacks will be posted to the application from PeerConnection. | 418 // All the callbacks will be posted to the application from PeerConnection. |
413 session_->RegisterIceObserver(this); | 419 session_->RegisterIceObserver(this); |
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799 break; | 805 break; |
800 } | 806 } |
801 } | 807 } |
802 | 808 |
803 void PeerConnection::OnAddRemoteStream(MediaStreamInterface* stream) { | 809 void PeerConnection::OnAddRemoteStream(MediaStreamInterface* stream) { |
804 stats_->AddStream(stream); | 810 stats_->AddStream(stream); |
805 observer_->OnAddStream(stream); | 811 observer_->OnAddStream(stream); |
806 } | 812 } |
807 | 813 |
808 void PeerConnection::OnRemoveRemoteStream(MediaStreamInterface* stream) { | 814 void PeerConnection::OnRemoveRemoteStream(MediaStreamInterface* stream) { |
809 stream_handler_container_->RemoveRemoteStream(stream); | |
810 observer_->OnRemoveStream(stream); | 815 observer_->OnRemoveStream(stream); |
811 } | 816 } |
812 | 817 |
813 void PeerConnection::OnAddDataChannel(DataChannelInterface* data_channel) { | 818 void PeerConnection::OnAddDataChannel(DataChannelInterface* data_channel) { |
814 observer_->OnDataChannel(DataChannelProxy::Create(signaling_thread(), | 819 observer_->OnDataChannel(DataChannelProxy::Create(signaling_thread(), |
815 data_channel)); | 820 data_channel)); |
816 } | 821 } |
817 | 822 |
818 void PeerConnection::OnAddRemoteAudioTrack(MediaStreamInterface* stream, | 823 void PeerConnection::OnAddRemoteAudioTrack(MediaStreamInterface* stream, |
819 AudioTrackInterface* audio_track, | 824 AudioTrackInterface* audio_track, |
820 uint32 ssrc) { | 825 uint32 ssrc, |
821 stream_handler_container_->AddRemoteAudioTrack(stream, audio_track, ssrc); | 826 const std::string& mid) { |
827 rtp_receivers_.push_back(rtc::scoped_refptr<RtpReceiverInterface>( | |
828 new rtc::RefCountedObject<AudioRtpReceiver>(audio_track, ssrc, mid, | |
829 session_.get()))); | |
822 } | 830 } |
823 | 831 |
824 void PeerConnection::OnAddRemoteVideoTrack(MediaStreamInterface* stream, | 832 void PeerConnection::OnAddRemoteVideoTrack(MediaStreamInterface* stream, |
825 VideoTrackInterface* video_track, | 833 VideoTrackInterface* video_track, |
826 uint32 ssrc) { | 834 uint32 ssrc, |
827 stream_handler_container_->AddRemoteVideoTrack(stream, video_track, ssrc); | 835 const std::string& mid) { |
836 rtp_receivers_.push_back(rtc::scoped_refptr<RtpReceiverInterface>( | |
837 new rtc::RefCountedObject<VideoRtpReceiver>(video_track, ssrc, mid, | |
838 session_.get()))); | |
828 } | 839 } |
829 | 840 |
830 void PeerConnection::OnRemoveRemoteAudioTrack( | 841 void PeerConnection::OnRemoveRemoteAudioTrack( |
831 MediaStreamInterface* stream, | 842 MediaStreamInterface* stream, |
832 AudioTrackInterface* audio_track) { | 843 AudioTrackInterface* audio_track) { |
833 stream_handler_container_->RemoveRemoteTrack(stream, audio_track); | 844 auto it = FindRtpReceiverForTrack(audio_track); |
845 if (it == rtp_receivers_.end()) { | |
846 LOG(LS_WARNING) << "RtpReceiver for track with id " << audio_track->id() | |
847 << " doesn't exist."; | |
848 } else { | |
849 (*it)->Stop(); | |
850 rtp_receivers_.erase(it); | |
pthatcher1
2015/09/17 04:25:46
Long-term, I think we want to keep senders and rec
Taylor Brandstetter
2015/09/23 00:10:45
Can't an m-line go away if someone calls "stop" on
pthatcher1
2015/09/23 14:47:39
No, an m-line can never go away.
| |
851 } | |
834 } | 852 } |
835 | 853 |
836 void PeerConnection::OnRemoveRemoteVideoTrack( | 854 void PeerConnection::OnRemoveRemoteVideoTrack( |
837 MediaStreamInterface* stream, | 855 MediaStreamInterface* stream, |
838 VideoTrackInterface* video_track) { | 856 VideoTrackInterface* video_track) { |
839 stream_handler_container_->RemoveRemoteTrack(stream, video_track); | 857 auto it = FindRtpReceiverForTrack(video_track); |
858 if (it == rtp_receivers_.end()) { | |
859 LOG(LS_WARNING) << "RtpReceiver for track with id " << video_track->id() | |
860 << " doesn't exist."; | |
861 } else { | |
862 (*it)->Stop(); | |
863 rtp_receivers_.erase(it); | |
864 } | |
840 } | 865 } |
866 | |
841 void PeerConnection::OnAddLocalAudioTrack(MediaStreamInterface* stream, | 867 void PeerConnection::OnAddLocalAudioTrack(MediaStreamInterface* stream, |
842 AudioTrackInterface* audio_track, | 868 AudioTrackInterface* audio_track, |
843 uint32 ssrc) { | 869 uint32 ssrc, |
844 stream_handler_container_->AddLocalAudioTrack(stream, audio_track, ssrc); | 870 const std::string& mid) { |
871 rtp_senders_.push_back(rtc::scoped_refptr<RtpSenderInterface>( | |
872 new rtc::RefCountedObject<AudioRtpSender>(audio_track, ssrc, mid, | |
873 session_.get()))); | |
845 stats_->AddLocalAudioTrack(audio_track, ssrc); | 874 stats_->AddLocalAudioTrack(audio_track, ssrc); |
846 } | 875 } |
876 | |
847 void PeerConnection::OnAddLocalVideoTrack(MediaStreamInterface* stream, | 877 void PeerConnection::OnAddLocalVideoTrack(MediaStreamInterface* stream, |
848 VideoTrackInterface* video_track, | 878 VideoTrackInterface* video_track, |
849 uint32 ssrc) { | 879 uint32 ssrc, |
850 stream_handler_container_->AddLocalVideoTrack(stream, video_track, ssrc); | 880 const std::string& mid) { |
881 rtp_senders_.push_back(rtc::scoped_refptr<RtpSenderInterface>( | |
882 new rtc::RefCountedObject<VideoRtpSender>(video_track, ssrc, mid, | |
883 session_.get()))); | |
851 } | 884 } |
852 | 885 |
853 void PeerConnection::OnRemoveLocalAudioTrack(MediaStreamInterface* stream, | 886 void PeerConnection::OnRemoveLocalAudioTrack(MediaStreamInterface* stream, |
854 AudioTrackInterface* audio_track, | 887 AudioTrackInterface* audio_track, |
855 uint32 ssrc) { | 888 uint32 ssrc) { |
856 stream_handler_container_->RemoveLocalTrack(stream, audio_track); | 889 auto it = FindRtpSenderForTrack(audio_track); |
890 if (it == rtp_senders_.end()) { | |
891 LOG(LS_WARNING) << "RtpSender for track with id " << audio_track->id() | |
892 << " doesn't exist."; | |
893 return; | |
894 } else { | |
895 (*it)->Stop(); | |
896 rtp_senders_.erase(it); | |
897 } | |
857 stats_->RemoveLocalAudioTrack(audio_track, ssrc); | 898 stats_->RemoveLocalAudioTrack(audio_track, ssrc); |
858 } | 899 } |
859 | 900 |
860 void PeerConnection::OnRemoveLocalVideoTrack(MediaStreamInterface* stream, | 901 void PeerConnection::OnRemoveLocalVideoTrack(MediaStreamInterface* stream, |
861 VideoTrackInterface* video_track) { | 902 VideoTrackInterface* video_track) { |
862 stream_handler_container_->RemoveLocalTrack(stream, video_track); | 903 auto it = FindRtpSenderForTrack(video_track); |
904 if (it == rtp_senders_.end()) { | |
905 LOG(LS_WARNING) << "RtpSender for track with id " << video_track->id() | |
906 << " doesn't exist."; | |
907 return; | |
908 } else { | |
909 (*it)->Stop(); | |
910 rtp_senders_.erase(it); | |
911 } | |
863 } | 912 } |
864 | 913 |
865 void PeerConnection::OnRemoveLocalStream(MediaStreamInterface* stream) { | 914 void PeerConnection::OnRemoveLocalStream(MediaStreamInterface* stream) { |
866 stream_handler_container_->RemoveLocalStream(stream); | |
867 } | 915 } |
868 | 916 |
869 void PeerConnection::OnIceConnectionChange( | 917 void PeerConnection::OnIceConnectionChange( |
870 PeerConnectionInterface::IceConnectionState new_state) { | 918 PeerConnectionInterface::IceConnectionState new_state) { |
871 ASSERT(signaling_thread()->IsCurrent()); | 919 ASSERT(signaling_thread()->IsCurrent()); |
872 ice_connection_state_ = new_state; | 920 ice_connection_state_ = new_state; |
873 observer_->OnIceConnectionChange(ice_connection_state_); | 921 observer_->OnIceConnectionChange(ice_connection_state_); |
874 } | 922 } |
875 | 923 |
876 void PeerConnection::OnIceGatheringChange( | 924 void PeerConnection::OnIceGatheringChange( |
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906 observer_->OnIceConnectionChange(ice_connection_state_); | 954 observer_->OnIceConnectionChange(ice_connection_state_); |
907 if (ice_gathering_state_ != kIceGatheringComplete) { | 955 if (ice_gathering_state_ != kIceGatheringComplete) { |
908 ice_gathering_state_ = kIceGatheringComplete; | 956 ice_gathering_state_ = kIceGatheringComplete; |
909 observer_->OnIceGatheringChange(ice_gathering_state_); | 957 observer_->OnIceGatheringChange(ice_gathering_state_); |
910 } | 958 } |
911 } | 959 } |
912 observer_->OnSignalingChange(signaling_state_); | 960 observer_->OnSignalingChange(signaling_state_); |
913 observer_->OnStateChange(PeerConnectionObserver::kSignalingState); | 961 observer_->OnStateChange(PeerConnectionObserver::kSignalingState); |
914 } | 962 } |
915 | 963 |
964 PeerConnectionInterface::RtpSenders::iterator | |
965 PeerConnection::FindRtpSenderForTrack(MediaStreamTrackInterface* track) { | |
966 return std::find_if( | |
967 rtp_senders_.begin(), rtp_senders_.end(), | |
968 [track](const rtc::scoped_refptr<RtpSenderInterface>& sender) | |
969 -> bool { return sender->track() == track; }); | |
pthatcher1
2015/09/17 04:25:46
Is the "-> bool" necessary? I thought it could in
Taylor Brandstetter
2015/09/23 00:10:45
Oh, I guess so. Removed.
| |
970 } | |
971 PeerConnectionInterface::RtpReceivers::iterator | |
972 PeerConnection::FindRtpReceiverForTrack(MediaStreamTrackInterface* track) { | |
973 return std::find_if( | |
974 rtp_receivers_.begin(), rtp_receivers_.end(), | |
975 [track](const rtc::scoped_refptr<RtpReceiverInterface>& receiver) | |
976 -> bool { return receiver->track() == track; }); | |
977 } | |
978 | |
916 } // namespace webrtc | 979 } // namespace webrtc |
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