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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/logging.h" | 11 #include "base/logging.h" |
12 #include "base/metrics/field_trial.h" | 12 #include "base/metrics/field_trial.h" |
13 #include "base/strings/string_util.h" | 13 #include "base/strings/string_util.h" |
14 #include "base/strings/utf_string_conversions.h" | 14 #include "base/strings/utf_string_conversions.h" |
15 #include "base/synchronization/waitable_event.h" | 15 #include "base/synchronization/waitable_event.h" |
16 #include "content/common/media/media_stream_messages.h" | 16 #include "content/common/media/media_stream_messages.h" |
17 #include "content/public/common/content_client.h" | 17 #include "content/public/common/content_client.h" |
18 #include "content/public/common/content_switches.h" | 18 #include "content/public/common/content_switches.h" |
19 #include "content/public/common/renderer_preferences.h" | 19 #include "content/public/common/renderer_preferences.h" |
20 #include "content/public/renderer/content_renderer_client.h" | 20 #include "content/public/renderer/content_renderer_client.h" |
21 #include "content/public/renderer/webrtc_ip_handling_policy.h" | |
21 #include "content/renderer/media/media_stream.h" | 22 #include "content/renderer/media/media_stream.h" |
22 #include "content/renderer/media/media_stream_audio_processor.h" | 23 #include "content/renderer/media/media_stream_audio_processor.h" |
23 #include "content/renderer/media/media_stream_audio_processor_options.h" | 24 #include "content/renderer/media/media_stream_audio_processor_options.h" |
24 #include "content/renderer/media/media_stream_audio_source.h" | 25 #include "content/renderer/media/media_stream_audio_source.h" |
25 #include "content/renderer/media/media_stream_video_source.h" | 26 #include "content/renderer/media/media_stream_video_source.h" |
26 #include "content/renderer/media/media_stream_video_track.h" | 27 #include "content/renderer/media/media_stream_video_track.h" |
27 #include "content/renderer/media/peer_connection_identity_store.h" | 28 #include "content/renderer/media/peer_connection_identity_store.h" |
28 #include "content/renderer/media/rtc_media_constraints.h" | 29 #include "content/renderer/media/rtc_media_constraints.h" |
29 #include "content/renderer/media/rtc_peer_connection_handler.h" | 30 #include "content/renderer/media/rtc_peer_connection_handler.h" |
30 #include "content/renderer/media/rtc_video_decoder_factory.h" | 31 #include "content/renderer/media/rtc_video_decoder_factory.h" |
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55 #include "third_party/WebKit/public/platform/WebURL.h" | 56 #include "third_party/WebKit/public/platform/WebURL.h" |
56 #include "third_party/WebKit/public/web/WebDocument.h" | 57 #include "third_party/WebKit/public/web/WebDocument.h" |
57 #include "third_party/WebKit/public/web/WebFrame.h" | 58 #include "third_party/WebKit/public/web/WebFrame.h" |
58 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface .h" | 59 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface .h" |
59 #include "third_party/webrtc/base/ssladapter.h" | 60 #include "third_party/webrtc/base/ssladapter.h" |
60 | 61 |
61 #if defined(OS_ANDROID) | 62 #if defined(OS_ANDROID) |
62 #include "media/base/android/media_codec_bridge.h" | 63 #include "media/base/android/media_codec_bridge.h" |
63 #endif | 64 #endif |
64 | 65 |
66 namespace { | |
67 | |
68 enum WebRTCIPPermissionMode { | |
pthatcher1
2015/10/21 00:06:53
It was called IPHandling in one place and IPPermis
guoweis_left_chromium
2015/10/23 16:40:49
Renamed it to be WebRTCIPHandlingPolicy to be cons
| |
69 DEFAULT, | |
70 DEFAULT_PUBLIC_INTERFACE_ONLY, | |
71 DISABLE_NON_PROXIED_UDP, | |
72 }; | |
73 | |
74 WebRTCIPPermissionMode GetWebRTCIPHandlingPolicy( | |
75 const std::string& preference) { | |
76 if (preference == content::kWebRTCIPHandlingDefaultPublicInterfaceOnly) | |
77 return DEFAULT_PUBLIC_INTERFACE_ONLY; | |
78 if (preference == content::kWebRTCIPHandlingDisableNonProxiedUdp) | |
79 return DISABLE_NON_PROXIED_UDP; | |
80 return DEFAULT; | |
81 } | |
82 | |
83 } // namespace | |
84 | |
65 namespace content { | 85 namespace content { |
66 | 86 |
67 // Map of corresponding media constraints and platform effects. | 87 // Map of corresponding media constraints and platform effects. |
68 struct { | 88 struct { |
69 const char* constraint; | 89 const char* constraint; |
70 const media::AudioParameters::PlatformEffectsMask effect; | 90 const media::AudioParameters::PlatformEffectsMask effect; |
71 } const kConstraintEffectMap[] = { | 91 } const kConstraintEffectMap[] = { |
72 { webrtc::MediaConstraintsInterface::kGoogEchoCancellation, | 92 { webrtc::MediaConstraintsInterface::kGoogEchoCancellation, |
73 media::AudioParameters::ECHO_CANCELLER }, | 93 media::AudioParameters::ECHO_CANCELLER }, |
74 }; | 94 }; |
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448 VLOG(3) << "WebRTC routing preferences will not be enforced"; | 468 VLOG(3) << "WebRTC routing preferences will not be enforced"; |
449 } else { | 469 } else { |
450 if (web_frame && web_frame->view()) { | 470 if (web_frame && web_frame->view()) { |
451 RenderViewImpl* renderer_view_impl = | 471 RenderViewImpl* renderer_view_impl = |
452 RenderViewImpl::FromWebView(web_frame->view()); | 472 RenderViewImpl::FromWebView(web_frame->view()); |
453 if (renderer_view_impl) { | 473 if (renderer_view_impl) { |
454 // TODO(guoweis): |enable_multiple_routes| should be renamed to | 474 // TODO(guoweis): |enable_multiple_routes| should be renamed to |
455 // |request_multiple_routes|. Whether local IP addresses could be | 475 // |request_multiple_routes|. Whether local IP addresses could be |
456 // collected depends on if mic/camera permission is granted for this | 476 // collected depends on if mic/camera permission is granted for this |
457 // origin. | 477 // origin. |
458 port_config.enable_multiple_routes = | 478 WebRTCIPPermissionMode mode = |
459 renderer_view_impl->renderer_preferences() | 479 GetWebRTCIPHandlingPolicy(renderer_view_impl->renderer_preferences() |
460 .enable_webrtc_multiple_routes; | 480 .webrtc_ip_handling_policy); |
461 port_config.enable_nonproxied_udp = | 481 |
462 renderer_view_impl->renderer_preferences() | 482 port_config.enable_multiple_routes = (mode == DEFAULT); |
463 .enable_webrtc_nonproxied_udp; | 483 port_config.enable_nonproxied_udp = (mode != DISABLE_NON_PROXIED_UDP); |
pthatcher1
2015/10/21 00:06:53
I think this might be more clear as:
switch(mode)
guoweis_left_chromium
2015/10/23 16:40:49
Done.
| |
464 VLOG(3) << "WebRTC routing preferences: multiple_routes: " | 484 VLOG(3) << "WebRTC routing preferences: multiple_routes: " |
465 << port_config.enable_multiple_routes | 485 << port_config.enable_multiple_routes |
466 << ", nonproxied_udp: " << port_config.enable_nonproxied_udp; | 486 << ", nonproxied_udp: " << port_config.enable_nonproxied_udp; |
467 } | 487 } |
468 } | 488 } |
469 if (port_config.enable_multiple_routes) { | 489 if (port_config.enable_multiple_routes) { |
470 bool create_media_permission = | 490 bool create_media_permission = |
471 base::CommandLine::ForCurrentProcess()->HasSwitch( | 491 base::CommandLine::ForCurrentProcess()->HasSwitch( |
472 switches::kEnforceWebRtcIPPermissionCheck); | 492 switches::kEnforceWebRtcIPPermissionCheck); |
473 create_media_permission = | 493 create_media_permission = |
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755 } | 775 } |
756 | 776 |
757 void PeerConnectionDependencyFactory::EnsureWebRtcAudioDeviceImpl() { | 777 void PeerConnectionDependencyFactory::EnsureWebRtcAudioDeviceImpl() { |
758 if (audio_device_.get()) | 778 if (audio_device_.get()) |
759 return; | 779 return; |
760 | 780 |
761 audio_device_ = new WebRtcAudioDeviceImpl(); | 781 audio_device_ = new WebRtcAudioDeviceImpl(); |
762 } | 782 } |
763 | 783 |
764 } // namespace content | 784 } // namespace content |
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