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1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. | 1 // Copyright (c) 2012 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/media_stream_dependency_factory.h" | 5 #include "content/renderer/media/media_stream_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/synchronization/waitable_event.h" | 10 #include "base/synchronization/waitable_event.h" |
11 #include "base/utf_string_conversions.h" | 11 #include "base/utf_string_conversions.h" |
12 #include "content/public/common/content_switches.h" | 12 #include "content/public/common/content_switches.h" |
| 13 #include "content/public/renderer/content_renderer_client.h" |
13 #include "content/renderer/media/media_stream_source_extra_data.h" | 14 #include "content/renderer/media/media_stream_source_extra_data.h" |
14 #include "content/renderer/media/rtc_media_constraints.h" | 15 #include "content/renderer/media/rtc_media_constraints.h" |
15 #include "content/renderer/media/rtc_peer_connection_handler.h" | 16 #include "content/renderer/media/rtc_peer_connection_handler.h" |
16 #include "content/renderer/media/rtc_video_capturer.h" | 17 #include "content/renderer/media/rtc_video_capturer.h" |
17 #include "content/renderer/media/video_capture_impl_manager.h" | 18 #include "content/renderer/media/video_capture_impl_manager.h" |
18 #include "content/renderer/media/webaudio_capturer_source.h" | 19 #include "content/renderer/media/webaudio_capturer_source.h" |
19 #include "content/renderer/media/webrtc_audio_device_impl.h" | 20 #include "content/renderer/media/webrtc_audio_device_impl.h" |
20 #include "content/renderer/media/webrtc_local_audio_track.h" | 21 #include "content/renderer/media/webrtc_local_audio_track.h" |
21 #include "content/renderer/media/webrtc_logging_handler_impl.h" | |
22 #include "content/renderer/media/webrtc_logging_message_filter.h" | |
23 #include "content/renderer/media/webrtc_uma_histograms.h" | 22 #include "content/renderer/media/webrtc_uma_histograms.h" |
24 #include "content/renderer/p2p/ipc_network_manager.h" | 23 #include "content/renderer/p2p/ipc_network_manager.h" |
25 #include "content/renderer/p2p/ipc_socket_factory.h" | 24 #include "content/renderer/p2p/ipc_socket_factory.h" |
26 #include "content/renderer/p2p/port_allocator.h" | 25 #include "content/renderer/p2p/port_allocator.h" |
27 #include "content/renderer/render_thread_impl.h" | 26 #include "content/renderer/render_thread_impl.h" |
28 #include "jingle/glue/thread_wrapper.h" | 27 #include "jingle/glue/thread_wrapper.h" |
29 #include "third_party/WebKit/Source/Platform/chromium/public/WebMediaConstraints
.h" | 28 #include "third_party/WebKit/Source/Platform/chromium/public/WebMediaConstraints
.h" |
30 #include "third_party/WebKit/Source/Platform/chromium/public/WebMediaStream.h" | 29 #include "third_party/WebKit/Source/Platform/chromium/public/WebMediaStream.h" |
31 #include "third_party/WebKit/Source/Platform/chromium/public/WebMediaStreamSourc
e.h" | 30 #include "third_party/WebKit/Source/Platform/chromium/public/WebMediaStreamSourc
e.h" |
32 #include "third_party/WebKit/Source/Platform/chromium/public/WebMediaStreamTrack
.h" | 31 #include "third_party/WebKit/Source/Platform/chromium/public/WebMediaStreamTrack
.h" |
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501 CHECK(web_frame); | 500 CHECK(web_frame); |
502 CHECK(observer); | 501 CHECK(observer); |
503 | 502 |
504 webrtc::MediaConstraintsInterface::Constraints optional_constraints = | 503 webrtc::MediaConstraintsInterface::Constraints optional_constraints = |
505 constraints->GetOptional(); | 504 constraints->GetOptional(); |
506 std::string constraint_value; | 505 std::string constraint_value; |
507 if (!webrtc_log_open_ && | 506 if (!webrtc_log_open_ && |
508 optional_constraints.FindFirst(kWebRtcLoggingConstraint, | 507 optional_constraints.FindFirst(kWebRtcLoggingConstraint, |
509 &constraint_value)) { | 508 &constraint_value)) { |
510 webrtc_log_open_ = true; | 509 webrtc_log_open_ = true; |
511 | 510 GetContentClient()->renderer()->InitWebRtcLogging(constraint_value); |
512 RenderThreadImpl::current()->GetIOMessageLoopProxy()->PostTask( | |
513 FROM_HERE, base::Bind( | |
514 &MediaStreamDependencyFactory::CreateWebRtcLoggingHandler, | |
515 base::Unretained(this), | |
516 RenderThreadImpl::current()->webrtc_logging_message_filter(), | |
517 constraint_value)); | |
518 } | 511 } |
519 | 512 |
520 scoped_refptr<P2PPortAllocatorFactory> pa_factory = | 513 scoped_refptr<P2PPortAllocatorFactory> pa_factory = |
521 new talk_base::RefCountedObject<P2PPortAllocatorFactory>( | 514 new talk_base::RefCountedObject<P2PPortAllocatorFactory>( |
522 p2p_socket_dispatcher_.get(), | 515 p2p_socket_dispatcher_.get(), |
523 network_manager_, | 516 network_manager_, |
524 socket_factory_.get(), | 517 socket_factory_.get(), |
525 web_frame); | 518 web_frame); |
526 return pc_factory_->CreatePeerConnection( | 519 return pc_factory_->CreatePeerConnection( |
527 ice_servers, constraints, pa_factory, observer).get(); | 520 ice_servers, constraints, pa_factory, observer).get(); |
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780 // Stopping the thread will wait until all tasks have been | 773 // Stopping the thread will wait until all tasks have been |
781 // processed before returning. We wait for the above task to finish before | 774 // processed before returning. We wait for the above task to finish before |
782 // letting the the function continue to avoid any potential race issues. | 775 // letting the the function continue to avoid any potential race issues. |
783 chrome_worker_thread_.Stop(); | 776 chrome_worker_thread_.Stop(); |
784 } else { | 777 } else { |
785 NOTREACHED() << "Worker thread not running."; | 778 NOTREACHED() << "Worker thread not running."; |
786 } | 779 } |
787 } | 780 } |
788 } | 781 } |
789 | 782 |
790 void MediaStreamDependencyFactory::CreateWebRtcLoggingHandler( | |
791 WebRtcLoggingMessageFilter* filter, | |
792 const std::string& app_session_id) { | |
793 WebRtcLoggingHandlerImpl* handler = | |
794 new WebRtcLoggingHandlerImpl(filter->io_message_loop()); | |
795 | |
796 // TODO(grunell): Give app session id as parameter. | |
797 filter->InitLogging(handler); | |
798 } | |
799 | |
800 } // namespace content | 783 } // namespace content |
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