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Side by Side Diff: content/renderer/media/webrtc/peer_connection_dependency_factory.cc

Issue 1335023002: Add UMA metrics and finch experiment for DTLS1.2 in WebRTC. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 5 years, 2 months ago
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1 // Copyright 2014 The Chromium Authors. All rights reserved. 1 // Copyright 2014 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 "content/renderer/media/webrtc/peer_connection_dependency_factory.h" 5 #include "content/renderer/media/webrtc/peer_connection_dependency_factory.h"
6 6
7 #include <vector> 7 #include <vector>
8 8
9 #include "base/command_line.h" 9 #include "base/command_line.h"
10 #include "base/location.h" 10 #include "base/location.h"
11 #include "base/metrics/field_trial.h"
11 #include "base/strings/utf_string_conversions.h" 12 #include "base/strings/utf_string_conversions.h"
12 #include "base/synchronization/waitable_event.h" 13 #include "base/synchronization/waitable_event.h"
13 #include "content/common/media/media_stream_messages.h" 14 #include "content/common/media/media_stream_messages.h"
14 #include "content/public/common/content_switches.h" 15 #include "content/public/common/content_switches.h"
15 #include "content/public/common/renderer_preferences.h" 16 #include "content/public/common/renderer_preferences.h"
16 #include "content/renderer/media/media_stream.h" 17 #include "content/renderer/media/media_stream.h"
17 #include "content/renderer/media/media_stream_audio_processor.h" 18 #include "content/renderer/media/media_stream_audio_processor.h"
18 #include "content/renderer/media/media_stream_audio_processor_options.h" 19 #include "content/renderer/media/media_stream_audio_processor_options.h"
19 #include "content/renderer/media/media_stream_audio_source.h" 20 #include "content/renderer/media/media_stream_audio_source.h"
20 #include "content/renderer/media/media_stream_video_source.h" 21 #include "content/renderer/media/media_stream_video_source.h"
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368 369
369 pc_factory_ = webrtc::CreatePeerConnectionFactory( 370 pc_factory_ = webrtc::CreatePeerConnectionFactory(
370 worker_thread_, signaling_thread_, audio_device_.get(), 371 worker_thread_, signaling_thread_, audio_device_.get(),
371 encoder_factory.release(), decoder_factory.release()); 372 encoder_factory.release(), decoder_factory.release());
372 CHECK(pc_factory_.get()); 373 CHECK(pc_factory_.get());
373 374
374 webrtc::PeerConnectionFactoryInterface::Options factory_options; 375 webrtc::PeerConnectionFactoryInterface::Options factory_options;
375 factory_options.disable_sctp_data_channels = false; 376 factory_options.disable_sctp_data_channels = false;
376 factory_options.disable_encryption = 377 factory_options.disable_encryption =
377 cmd_line->HasSwitch(switches::kDisableWebRtcEncryption); 378 cmd_line->HasSwitch(switches::kDisableWebRtcEncryption);
378 if (cmd_line->HasSwitch(switches::kEnableWebRtcDtls12)) 379 if (cmd_line->HasSwitch(switches::kEnableWebRtcDtls12) ||
380 base::FieldTrialList::FindFullName("WebRTC-PeerConnectionDTLS1.2") ==
381 "Enabled") {
379 factory_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12; 382 factory_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
383 }
380 pc_factory_->SetOptions(factory_options); 384 pc_factory_->SetOptions(factory_options);
381 385
382 event->Signal(); 386 event->Signal();
383 } 387 }
384 388
385 bool PeerConnectionDependencyFactory::PeerConnectionFactoryCreated() { 389 bool PeerConnectionDependencyFactory::PeerConnectionFactoryCreated() {
386 return pc_factory_.get() != NULL; 390 return pc_factory_.get() != NULL;
387 } 391 }
388 392
389 scoped_refptr<webrtc::PeerConnectionInterface> 393 scoped_refptr<webrtc::PeerConnectionInterface>
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681 } 685 }
682 686
683 void PeerConnectionDependencyFactory::EnsureWebRtcAudioDeviceImpl() { 687 void PeerConnectionDependencyFactory::EnsureWebRtcAudioDeviceImpl() {
684 if (audio_device_.get()) 688 if (audio_device_.get())
685 return; 689 return;
686 690
687 audio_device_ = new WebRtcAudioDeviceImpl(); 691 audio_device_ = new WebRtcAudioDeviceImpl();
688 } 692 }
689 693
690 } // namespace content 694 } // namespace content
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