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Side by Side Diff: talk/app/webrtc/peerconnection.cc

Issue 1351803002: Exposing RtpSenders and RtpReceivers from PeerConnection. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Replacing "DetachFromProvider" with "Stop" and some TODOs. Created 5 years, 2 months ago
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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,
(...skipping 15 matching lines...) Expand all
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 : senders_) {
349 sender->Stop();
350 }
351 for (const auto& receiver : 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)
(...skipping 36 matching lines...) Expand 10 before | Expand all | Expand 10 after
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);
414 session_->SignalState.connect(this, &PeerConnection::OnSessionStateChange); 420 session_->SignalState.connect(this, &PeerConnection::OnSessionStateChange);
415 return true; 421 return true;
416 } 422 }
417 423
418 rtc::scoped_refptr<StreamCollectionInterface> 424 rtc::scoped_refptr<StreamCollectionInterface>
419 PeerConnection::local_streams() { 425 PeerConnection::local_streams() {
420 return mediastream_signaling_->local_streams(); 426 return mediastream_signaling_->local_streams();
421 } 427 }
422 428
423 rtc::scoped_refptr<StreamCollectionInterface> 429 rtc::scoped_refptr<StreamCollectionInterface>
424 PeerConnection::remote_streams() { 430 PeerConnection::remote_streams() {
425 return mediastream_signaling_->remote_streams(); 431 return mediastream_signaling_->remote_streams();
426 } 432 }
427 433
434 // TODO(deadbeef): Create RtpSenders immediately here, even if local
435 // description hasn't yet been set.
428 bool PeerConnection::AddStream(MediaStreamInterface* local_stream) { 436 bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
429 if (IsClosed()) { 437 if (IsClosed()) {
430 return false; 438 return false;
431 } 439 }
432 if (!CanAddLocalMediaStream(mediastream_signaling_->local_streams(), 440 if (!CanAddLocalMediaStream(mediastream_signaling_->local_streams(),
433 local_stream)) 441 local_stream))
434 return false; 442 return false;
435 443
436 if (!mediastream_signaling_->AddLocalStream(local_stream)) { 444 if (!mediastream_signaling_->AddLocalStream(local_stream)) {
437 return false; 445 return false;
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462 470
463 rtc::scoped_refptr<DtmfSenderInterface> sender( 471 rtc::scoped_refptr<DtmfSenderInterface> sender(
464 DtmfSender::Create(track, signaling_thread(), session_.get())); 472 DtmfSender::Create(track, signaling_thread(), session_.get()));
465 if (!sender.get()) { 473 if (!sender.get()) {
466 LOG(LS_ERROR) << "CreateDtmfSender failed on DtmfSender::Create."; 474 LOG(LS_ERROR) << "CreateDtmfSender failed on DtmfSender::Create.";
467 return NULL; 475 return NULL;
468 } 476 }
469 return DtmfSenderProxy::Create(signaling_thread(), sender.get()); 477 return DtmfSenderProxy::Create(signaling_thread(), sender.get());
470 } 478 }
471 479
480 std::vector<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::GetSenders()
481 const {
482 std::vector<rtc::scoped_refptr<RtpSenderInterface>> senders;
483 for (const auto& sender : senders_) {
484 senders.push_back(RtpSenderProxy::Create(signaling_thread(), sender.get()));
485 }
486 return senders;
487 }
488
489 std::vector<rtc::scoped_refptr<RtpReceiverInterface>>
490 PeerConnection::GetReceivers() const {
491 std::vector<rtc::scoped_refptr<RtpReceiverInterface>> receivers;
492 for (const auto& receiver : receivers_) {
493 receivers.push_back(
494 RtpReceiverProxy::Create(signaling_thread(), receiver.get()));
495 }
496 return receivers;
497 }
498
472 bool PeerConnection::GetStats(StatsObserver* observer, 499 bool PeerConnection::GetStats(StatsObserver* observer,
473 MediaStreamTrackInterface* track, 500 MediaStreamTrackInterface* track,
474 StatsOutputLevel level) { 501 StatsOutputLevel level) {
475 ASSERT(signaling_thread()->IsCurrent()); 502 ASSERT(signaling_thread()->IsCurrent());
476 if (!VERIFY(observer != NULL)) { 503 if (!VERIFY(observer != NULL)) {
477 LOG(LS_ERROR) << "GetStats - observer is NULL."; 504 LOG(LS_ERROR) << "GetStats - observer is NULL.";
478 return false; 505 return false;
479 } 506 }
480 507
481 stats_->UpdateStats(level); 508 stats_->UpdateStats(level);
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799 break; 826 break;
800 } 827 }
801 } 828 }
802 829
803 void PeerConnection::OnAddRemoteStream(MediaStreamInterface* stream) { 830 void PeerConnection::OnAddRemoteStream(MediaStreamInterface* stream) {
804 stats_->AddStream(stream); 831 stats_->AddStream(stream);
805 observer_->OnAddStream(stream); 832 observer_->OnAddStream(stream);
806 } 833 }
807 834
808 void PeerConnection::OnRemoveRemoteStream(MediaStreamInterface* stream) { 835 void PeerConnection::OnRemoveRemoteStream(MediaStreamInterface* stream) {
809 stream_handler_container_->RemoveRemoteStream(stream);
810 observer_->OnRemoveStream(stream); 836 observer_->OnRemoveStream(stream);
811 } 837 }
812 838
813 void PeerConnection::OnAddDataChannel(DataChannelInterface* data_channel) { 839 void PeerConnection::OnAddDataChannel(DataChannelInterface* data_channel) {
814 observer_->OnDataChannel(DataChannelProxy::Create(signaling_thread(), 840 observer_->OnDataChannel(DataChannelProxy::Create(signaling_thread(),
815 data_channel)); 841 data_channel));
816 } 842 }
817 843
818 void PeerConnection::OnAddRemoteAudioTrack(MediaStreamInterface* stream, 844 void PeerConnection::OnAddRemoteAudioTrack(MediaStreamInterface* stream,
819 AudioTrackInterface* audio_track, 845 AudioTrackInterface* audio_track,
820 uint32 ssrc) { 846 uint32 ssrc) {
821 stream_handler_container_->AddRemoteAudioTrack(stream, audio_track, ssrc); 847 receivers_.push_back(new AudioRtpReceiver(audio_track, ssrc, session_.get()));
822 } 848 }
823 849
824 void PeerConnection::OnAddRemoteVideoTrack(MediaStreamInterface* stream, 850 void PeerConnection::OnAddRemoteVideoTrack(MediaStreamInterface* stream,
825 VideoTrackInterface* video_track, 851 VideoTrackInterface* video_track,
826 uint32 ssrc) { 852 uint32 ssrc) {
827 stream_handler_container_->AddRemoteVideoTrack(stream, video_track, ssrc); 853 receivers_.push_back(new VideoRtpReceiver(video_track, ssrc, session_.get()));
828 } 854 }
829 855
856 // TODO(deadbeef): Keep RtpReceivers around even if track goes away in remote
857 // description.
830 void PeerConnection::OnRemoveRemoteAudioTrack( 858 void PeerConnection::OnRemoveRemoteAudioTrack(
831 MediaStreamInterface* stream, 859 MediaStreamInterface* stream,
832 AudioTrackInterface* audio_track) { 860 AudioTrackInterface* audio_track) {
833 stream_handler_container_->RemoveRemoteTrack(stream, audio_track); 861 auto it = FindReceiverForTrack(audio_track);
862 if (it == receivers_.end()) {
863 LOG(LS_WARNING) << "RtpReceiver for track with id " << audio_track->id()
864 << " doesn't exist.";
865 } else {
866 (*it)->Stop();
867 receivers_.erase(it);
868 }
834 } 869 }
835 870
836 void PeerConnection::OnRemoveRemoteVideoTrack( 871 void PeerConnection::OnRemoveRemoteVideoTrack(
837 MediaStreamInterface* stream, 872 MediaStreamInterface* stream,
838 VideoTrackInterface* video_track) { 873 VideoTrackInterface* video_track) {
839 stream_handler_container_->RemoveRemoteTrack(stream, video_track); 874 auto it = FindReceiverForTrack(video_track);
875 if (it == receivers_.end()) {
876 LOG(LS_WARNING) << "RtpReceiver for track with id " << video_track->id()
877 << " doesn't exist.";
878 } else {
879 (*it)->Stop();
880 receivers_.erase(it);
881 }
840 } 882 }
883
841 void PeerConnection::OnAddLocalAudioTrack(MediaStreamInterface* stream, 884 void PeerConnection::OnAddLocalAudioTrack(MediaStreamInterface* stream,
842 AudioTrackInterface* audio_track, 885 AudioTrackInterface* audio_track,
843 uint32 ssrc) { 886 uint32 ssrc) {
844 stream_handler_container_->AddLocalAudioTrack(stream, audio_track, ssrc); 887 senders_.push_back(new AudioRtpSender(audio_track, ssrc, session_.get()));
845 stats_->AddLocalAudioTrack(audio_track, ssrc); 888 stats_->AddLocalAudioTrack(audio_track, ssrc);
846 } 889 }
890
847 void PeerConnection::OnAddLocalVideoTrack(MediaStreamInterface* stream, 891 void PeerConnection::OnAddLocalVideoTrack(MediaStreamInterface* stream,
848 VideoTrackInterface* video_track, 892 VideoTrackInterface* video_track,
849 uint32 ssrc) { 893 uint32 ssrc) {
850 stream_handler_container_->AddLocalVideoTrack(stream, video_track, ssrc); 894 senders_.push_back(new VideoRtpSender(video_track, ssrc, session_.get()));
851 } 895 }
852 896
897 // TODO(deadbeef): Keep RtpSenders around even if track goes away in local
898 // description.
853 void PeerConnection::OnRemoveLocalAudioTrack(MediaStreamInterface* stream, 899 void PeerConnection::OnRemoveLocalAudioTrack(MediaStreamInterface* stream,
854 AudioTrackInterface* audio_track, 900 AudioTrackInterface* audio_track,
855 uint32 ssrc) { 901 uint32 ssrc) {
856 stream_handler_container_->RemoveLocalTrack(stream, audio_track); 902 auto it = FindSenderForTrack(audio_track);
903 if (it == senders_.end()) {
904 LOG(LS_WARNING) << "RtpSender for track with id " << audio_track->id()
905 << " doesn't exist.";
906 return;
907 } else {
908 (*it)->Stop();
909 senders_.erase(it);
910 }
857 stats_->RemoveLocalAudioTrack(audio_track, ssrc); 911 stats_->RemoveLocalAudioTrack(audio_track, ssrc);
858 } 912 }
859 913
860 void PeerConnection::OnRemoveLocalVideoTrack(MediaStreamInterface* stream, 914 void PeerConnection::OnRemoveLocalVideoTrack(MediaStreamInterface* stream,
861 VideoTrackInterface* video_track) { 915 VideoTrackInterface* video_track) {
862 stream_handler_container_->RemoveLocalTrack(stream, video_track); 916 auto it = FindSenderForTrack(video_track);
917 if (it == senders_.end()) {
918 LOG(LS_WARNING) << "RtpSender for track with id " << video_track->id()
919 << " doesn't exist.";
920 return;
921 } else {
922 (*it)->Stop();
923 senders_.erase(it);
924 }
863 } 925 }
864 926
865 void PeerConnection::OnRemoveLocalStream(MediaStreamInterface* stream) { 927 void PeerConnection::OnRemoveLocalStream(MediaStreamInterface* stream) {
866 stream_handler_container_->RemoveLocalStream(stream);
867 } 928 }
868 929
869 void PeerConnection::OnIceConnectionChange( 930 void PeerConnection::OnIceConnectionChange(
870 PeerConnectionInterface::IceConnectionState new_state) { 931 PeerConnectionInterface::IceConnectionState new_state) {
871 ASSERT(signaling_thread()->IsCurrent()); 932 ASSERT(signaling_thread()->IsCurrent());
872 ice_connection_state_ = new_state; 933 ice_connection_state_ = new_state;
873 observer_->OnIceConnectionChange(ice_connection_state_); 934 observer_->OnIceConnectionChange(ice_connection_state_);
874 } 935 }
875 936
876 void PeerConnection::OnIceGatheringChange( 937 void PeerConnection::OnIceGatheringChange(
(...skipping 29 matching lines...) Expand all
906 observer_->OnIceConnectionChange(ice_connection_state_); 967 observer_->OnIceConnectionChange(ice_connection_state_);
907 if (ice_gathering_state_ != kIceGatheringComplete) { 968 if (ice_gathering_state_ != kIceGatheringComplete) {
908 ice_gathering_state_ = kIceGatheringComplete; 969 ice_gathering_state_ = kIceGatheringComplete;
909 observer_->OnIceGatheringChange(ice_gathering_state_); 970 observer_->OnIceGatheringChange(ice_gathering_state_);
910 } 971 }
911 } 972 }
912 observer_->OnSignalingChange(signaling_state_); 973 observer_->OnSignalingChange(signaling_state_);
913 observer_->OnStateChange(PeerConnectionObserver::kSignalingState); 974 observer_->OnStateChange(PeerConnectionObserver::kSignalingState);
914 } 975 }
915 976
977 std::vector<rtc::scoped_refptr<RtpSenderInterface>>::iterator
978 PeerConnection::FindSenderForTrack(MediaStreamTrackInterface* track) {
979 return std::find_if(
980 senders_.begin(), senders_.end(),
981 [track](const rtc::scoped_refptr<RtpSenderInterface>& sender) {
982 return sender->track() == track;
983 });
984 }
985
986 std::vector<rtc::scoped_refptr<RtpReceiverInterface>>::iterator
987 PeerConnection::FindReceiverForTrack(MediaStreamTrackInterface* track) {
988 return std::find_if(
989 receivers_.begin(), receivers_.end(),
990 [track](const rtc::scoped_refptr<RtpReceiverInterface>& receiver) {
991 return receiver->track() == track;
992 });
993 }
994
916 } // namespace webrtc 995 } // namespace webrtc
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