Chromium Code Reviews| OLD | NEW |
|---|---|
| 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/strings/utf_string_conversions.h" | 10 #include "base/strings/utf_string_conversions.h" |
| (...skipping 532 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
| 543 url)); | 543 url)); |
| 544 } | 544 } |
| 545 | 545 |
| 546 scoped_refptr<P2PPortAllocatorFactory> pa_factory = | 546 scoped_refptr<P2PPortAllocatorFactory> pa_factory = |
| 547 new talk_base::RefCountedObject<P2PPortAllocatorFactory>( | 547 new talk_base::RefCountedObject<P2PPortAllocatorFactory>( |
| 548 p2p_socket_dispatcher_.get(), | 548 p2p_socket_dispatcher_.get(), |
| 549 network_manager_, | 549 network_manager_, |
| 550 socket_factory_.get(), | 550 socket_factory_.get(), |
| 551 web_frame); | 551 web_frame); |
| 552 | 552 |
| 553 // The crypto APIs needed for generating identities are not implenented for | |
| 554 // OPENSSL yet. So passing NULL to let Libjingle generate its own. | |
| 555 // TODO(jiayl): remove the #if once the crypto APIs are implemented for OPENSSL. | |
| 553 PeerConnectionIdentityService* identity_service = | 556 PeerConnectionIdentityService* identity_service = |
| 557 #if !defined(USE_OPENSSL) | |
|
Ami GONE FROM CHROMIUM
2013/08/09 17:14:35
I don't really understand this. Would it be usefu
jiayl
2013/08/09 17:41:16
Added the link to the existing bug.
I feel it's a
| |
| 554 new PeerConnectionIdentityService(GURL(web_frame->document().url().spec()) | 558 new PeerConnectionIdentityService(GURL(web_frame->document().url().spec()) |
| 555 .GetOrigin()); | 559 .GetOrigin()); |
| 560 #else | |
| 561 NULL; | |
| 562 #endif // !defined(USE_OPENSSL) | |
| 556 | 563 |
| 557 return pc_factory_->CreatePeerConnection(ice_servers, | 564 return pc_factory_->CreatePeerConnection(ice_servers, |
| 558 constraints, | 565 constraints, |
| 559 pa_factory.get(), | 566 pa_factory.get(), |
| 560 identity_service, | 567 identity_service, |
| 561 observer).get(); | 568 observer).get(); |
| 562 } | 569 } |
| 563 | 570 |
| 564 scoped_refptr<webrtc::MediaStreamInterface> | 571 scoped_refptr<webrtc::MediaStreamInterface> |
| 565 MediaStreamDependencyFactory::CreateLocalMediaStream( | 572 MediaStreamDependencyFactory::CreateLocalMediaStream( |
| (...skipping 250 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
| 816 } | 823 } |
| 817 | 824 |
| 818 // Add the capturer to the WebRtcAudioDeviceImpl if it is a new capturer. | 825 // Add the capturer to the WebRtcAudioDeviceImpl if it is a new capturer. |
| 819 if (is_new_capturer) | 826 if (is_new_capturer) |
| 820 GetWebRtcAudioDevice()->AddAudioCapturer(capturer); | 827 GetWebRtcAudioDevice()->AddAudioCapturer(capturer); |
| 821 | 828 |
| 822 return capturer; | 829 return capturer; |
| 823 } | 830 } |
| 824 | 831 |
| 825 } // namespace content | 832 } // namespace content |
| OLD | NEW |