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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"
12 #include "base/strings/string_util.h"
11 #include "base/strings/utf_string_conversions.h" 13 #include "base/strings/utf_string_conversions.h"
12 #include "base/synchronization/waitable_event.h" 14 #include "base/synchronization/waitable_event.h"
13 #include "content/common/media/media_stream_messages.h" 15 #include "content/common/media/media_stream_messages.h"
14 #include "content/public/common/content_switches.h" 16 #include "content/public/common/content_switches.h"
15 #include "content/public/common/renderer_preferences.h" 17 #include "content/public/common/renderer_preferences.h"
16 #include "content/renderer/media/media_stream.h" 18 #include "content/renderer/media/media_stream.h"
17 #include "content/renderer/media/media_stream_audio_processor.h" 19 #include "content/renderer/media/media_stream_audio_processor.h"
18 #include "content/renderer/media/media_stream_audio_processor_options.h" 20 #include "content/renderer/media/media_stream_audio_processor_options.h"
19 #include "content/renderer/media/media_stream_audio_source.h" 21 #include "content/renderer/media/media_stream_audio_source.h"
20 #include "content/renderer/media/media_stream_video_source.h" 22 #include "content/renderer/media/media_stream_video_source.h"
(...skipping 347 matching lines...) Expand 10 before | Expand all | Expand 10 after
368 370
369 pc_factory_ = webrtc::CreatePeerConnectionFactory( 371 pc_factory_ = webrtc::CreatePeerConnectionFactory(
370 worker_thread_, signaling_thread_, audio_device_.get(), 372 worker_thread_, signaling_thread_, audio_device_.get(),
371 encoder_factory.release(), decoder_factory.release()); 373 encoder_factory.release(), decoder_factory.release());
372 CHECK(pc_factory_.get()); 374 CHECK(pc_factory_.get());
373 375
374 webrtc::PeerConnectionFactoryInterface::Options factory_options; 376 webrtc::PeerConnectionFactoryInterface::Options factory_options;
375 factory_options.disable_sctp_data_channels = false; 377 factory_options.disable_sctp_data_channels = false;
376 factory_options.disable_encryption = 378 factory_options.disable_encryption =
377 cmd_line->HasSwitch(switches::kDisableWebRtcEncryption); 379 cmd_line->HasSwitch(switches::kDisableWebRtcEncryption);
378 if (cmd_line->HasSwitch(switches::kEnableWebRtcDtls12)) 380
381 std::string group_name =
382 base::FieldTrialList::FindFullName("WebRTC-PeerConnectionDTLS1.2");
383 if (cmd_line->HasSwitch(switches::kEnableWebRtcDtls12) ||
384 StartsWith(group_name, "Enabled", base::CompareCase::SENSITIVE)) {
tommi (sloooow) - chröme 2015/09/30 19:24:45 nit: StartsWith() check first since you've already
379 factory_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12; 385 factory_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
386 }
380 pc_factory_->SetOptions(factory_options); 387 pc_factory_->SetOptions(factory_options);
381 388
382 event->Signal(); 389 event->Signal();
383 } 390 }
384 391
385 bool PeerConnectionDependencyFactory::PeerConnectionFactoryCreated() { 392 bool PeerConnectionDependencyFactory::PeerConnectionFactoryCreated() {
386 return pc_factory_.get() != NULL; 393 return pc_factory_.get() != NULL;
387 } 394 }
388 395
389 scoped_refptr<webrtc::PeerConnectionInterface> 396 scoped_refptr<webrtc::PeerConnectionInterface>
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681 } 688 }
682 689
683 void PeerConnectionDependencyFactory::EnsureWebRtcAudioDeviceImpl() { 690 void PeerConnectionDependencyFactory::EnsureWebRtcAudioDeviceImpl() {
684 if (audio_device_.get()) 691 if (audio_device_.get())
685 return; 692 return;
686 693
687 audio_device_ = new WebRtcAudioDeviceImpl(); 694 audio_device_ = new WebRtcAudioDeviceImpl();
688 } 695 }
689 696
690 } // namespace content 697 } // namespace content
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