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| 1 // Copyright 2015 The Chromium Authors. All rights reserved. | 1 // Copyright 2015 The Chromium Authors. All rights reserved. |
| 2 // Use of this source code is governed by a BSD-style license that can be | 2 // Use of this source code is governed by a BSD-style license that can be |
| 3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
| 4 | 4 |
| 5 #include "remoting/protocol/webrtc_transport.h" | 5 #include "remoting/protocol/webrtc_transport.h" |
| 6 | 6 |
| 7 #include "base/callback_helpers.h" | 7 #include "base/callback_helpers.h" |
| 8 #include "base/single_thread_task_runner.h" | 8 #include "base/single_thread_task_runner.h" |
| 9 #include "base/strings/string_number_conversions.h" | 9 #include "base/strings/string_number_conversions.h" |
| 10 #include "base/task_runner_util.h" | 10 #include "base/task_runner_util.h" |
| 11 #include "jingle/glue/thread_wrapper.h" | 11 #include "jingle/glue/thread_wrapper.h" |
| 12 #include "remoting/protocol/chromium_port_allocator.h" | |
| 12 #include "third_party/libjingle/source/talk/app/webrtc/test/fakeconstraints.h" | 13 #include "third_party/libjingle/source/talk/app/webrtc/test/fakeconstraints.h" |
| 13 #include "third_party/webrtc/libjingle/xmllite/xmlelement.h" | 14 #include "third_party/webrtc/libjingle/xmllite/xmlelement.h" |
| 14 #include "third_party/webrtc/modules/audio_device/include/fake_audio_device.h" | 15 #include "third_party/webrtc/modules/audio_device/include/fake_audio_device.h" |
| 15 | 16 |
| 16 using buzz::QName; | 17 using buzz::QName; |
| 17 using buzz::XmlElement; | 18 using buzz::XmlElement; |
| 18 | 19 |
| 19 namespace remoting { | 20 namespace remoting { |
| 20 namespace protocol { | 21 namespace protocol { |
| 21 | 22 |
| (...skipping 79 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
| 101 | 102 |
| 102 private: | 103 private: |
| 103 ResultCallback result_callback_; | 104 ResultCallback result_callback_; |
| 104 | 105 |
| 105 DISALLOW_COPY_AND_ASSIGN(SetSessionDescriptionObserver); | 106 DISALLOW_COPY_AND_ASSIGN(SetSessionDescriptionObserver); |
| 106 }; | 107 }; |
| 107 | 108 |
| 108 } // namespace | 109 } // namespace |
| 109 | 110 |
| 110 WebrtcTransport::WebrtcTransport( | 111 WebrtcTransport::WebrtcTransport( |
| 111 rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface> | 112 const NetworkSettings& network_settings, |
|
Sergey Ulanov
2015/12/04 18:54:07
Pass PortAllocator instead of PortAllocatorFactory
| |
| 112 port_allocator_factory, | |
| 113 TransportRole role, | 113 TransportRole role, |
| 114 scoped_refptr<base::SingleThreadTaskRunner> worker_task_runner) | 114 scoped_refptr<base::SingleThreadTaskRunner> worker_task_runner) |
| 115 : port_allocator_factory_(port_allocator_factory), | 115 : network_settings_(network_settings), |
| 116 role_(role), | 116 role_(role), |
| 117 worker_task_runner_(worker_task_runner), | 117 worker_task_runner_(worker_task_runner), |
| 118 weak_factory_(this) {} | 118 weak_factory_(this) {} |
| 119 | 119 |
| 120 WebrtcTransport::~WebrtcTransport() {} | 120 WebrtcTransport::~WebrtcTransport() {} |
| 121 | 121 |
| 122 void WebrtcTransport::Start(EventHandler* event_handler, | 122 void WebrtcTransport::Start(EventHandler* event_handler, |
| 123 Authenticator* authenticator) { | 123 Authenticator* authenticator) { |
| 124 DCHECK(thread_checker_.CalledOnValidThread()); | 124 DCHECK(thread_checker_.CalledOnValidThread()); |
| 125 | 125 |
| (...skipping 117 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
| 243 | 243 |
| 244 webrtc::PeerConnectionInterface::IceServer stun_server; | 244 webrtc::PeerConnectionInterface::IceServer stun_server; |
| 245 stun_server.urls.push_back("stun:stun.l.google.com:19302"); | 245 stun_server.urls.push_back("stun:stun.l.google.com:19302"); |
| 246 webrtc::PeerConnectionInterface::RTCConfiguration rtc_config; | 246 webrtc::PeerConnectionInterface::RTCConfiguration rtc_config; |
| 247 rtc_config.servers.push_back(stun_server); | 247 rtc_config.servers.push_back(stun_server); |
| 248 | 248 |
| 249 webrtc::FakeConstraints constraints; | 249 webrtc::FakeConstraints constraints; |
| 250 constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, | 250 constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, |
| 251 webrtc::MediaConstraintsInterface::kValueTrue); | 251 webrtc::MediaConstraintsInterface::kValueTrue); |
| 252 | 252 |
| 253 rtc::scoped_ptr<ChromiumPortAllocator> port_allocator( | |
| 254 ChromiumPortAllocator::Create(nullptr, network_settings_).release()); | |
|
Sergey Ulanov
2015/12/04 18:54:07
This is not right. This class shouldn't depend on
| |
| 255 | |
| 253 peer_connection_ = peer_connection_factory_->CreatePeerConnection( | 256 peer_connection_ = peer_connection_factory_->CreatePeerConnection( |
| 254 rtc_config, &constraints, port_allocator_factory_, nullptr, this); | 257 rtc_config, &constraints, std::move(port_allocator), nullptr, this); |
| 255 | 258 |
| 256 data_stream_adapter_.Initialize(peer_connection_, | 259 data_stream_adapter_.Initialize(peer_connection_, |
| 257 role_ == TransportRole::SERVER); | 260 role_ == TransportRole::SERVER); |
| 258 | 261 |
| 259 if (role_ == TransportRole::CLIENT) { | 262 if (role_ == TransportRole::CLIENT) { |
| 260 webrtc::FakeConstraints offer_config; | 263 webrtc::FakeConstraints offer_config; |
| 261 offer_config.AddMandatory( | 264 offer_config.AddMandatory( |
| 262 webrtc::MediaConstraintsInterface::kOfferToReceiveVideo, | 265 webrtc::MediaConstraintsInterface::kOfferToReceiveVideo, |
| 263 webrtc::MediaConstraintsInterface::kValueTrue); | 266 webrtc::MediaConstraintsInterface::kValueTrue); |
| 264 offer_config.AddMandatory( | 267 offer_config.AddMandatory( |
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| 466 Close(INCOMPATIBLE_PROTOCOL); | 469 Close(INCOMPATIBLE_PROTOCOL); |
| 467 return; | 470 return; |
| 468 } | 471 } |
| 469 } | 472 } |
| 470 pending_incoming_candidates_.clear(); | 473 pending_incoming_candidates_.clear(); |
| 471 } | 474 } |
| 472 } | 475 } |
| 473 | 476 |
| 474 WebrtcTransportFactory::WebrtcTransportFactory( | 477 WebrtcTransportFactory::WebrtcTransportFactory( |
| 475 SignalStrategy* signal_strategy, | 478 SignalStrategy* signal_strategy, |
| 476 rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface> | 479 const NetworkSettings& network_settings, |
| 477 port_allocator_factory, | |
| 478 TransportRole role) | 480 TransportRole role) |
| 479 : signal_strategy_(signal_strategy), | 481 : signal_strategy_(signal_strategy), |
| 480 port_allocator_factory_(port_allocator_factory), | 482 network_settings_(network_settings), |
| 481 role_(role), | 483 role_(role), |
| 482 worker_thread_("ChromotingWebrtcWorkerThread") { | 484 worker_thread_("ChromotingWebrtcWorkerThread") { |
| 483 worker_thread_.StartWithOptions( | 485 worker_thread_.StartWithOptions( |
| 484 base::Thread::Options(base::MessageLoop::TYPE_IO, 0)); | 486 base::Thread::Options(base::MessageLoop::TYPE_IO, 0)); |
| 485 } | 487 } |
| 486 | 488 |
| 487 WebrtcTransportFactory::~WebrtcTransportFactory() {} | 489 WebrtcTransportFactory::~WebrtcTransportFactory() {} |
| 488 | 490 |
| 489 scoped_ptr<Transport> WebrtcTransportFactory::CreateTransport() { | 491 scoped_ptr<Transport> WebrtcTransportFactory::CreateTransport() { |
| 490 return make_scoped_ptr(new WebrtcTransport(port_allocator_factory_, role_, | 492 return make_scoped_ptr(new WebrtcTransport(network_settings_, role_, |
| 491 worker_thread_.task_runner())); | 493 worker_thread_.task_runner())); |
| 492 } | 494 } |
| 493 | 495 |
| 494 } // namespace protocol | 496 } // namespace protocol |
| 495 } // namespace remoting | 497 } // namespace remoting |
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