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Issue 1413393003: Change WebRTC IP handling policy from multiple booleans to an enum. (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 (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 { 5 {
6 'dependencies': [ 6 'dependencies': [
7 '../base/base.gyp:base', 7 '../base/base.gyp:base',
8 '../base/third_party/dynamic_annotations/dynamic_annotations.gyp:dynamic_ann otations', 8 '../base/third_party/dynamic_annotations/dynamic_annotations.gyp:dynamic_ann otations',
9 '../cc/cc.gyp:cc', 9 '../cc/cc.gyp:cc',
10 '../cc/blink/cc_blink.gyp:cc_blink', 10 '../cc/blink/cc_blink.gyp:cc_blink',
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601 ], 601 ],
602 'public_renderer_webrtc_sources': [ 602 'public_renderer_webrtc_sources': [
603 'public/renderer/media_stream_api.cc', 603 'public/renderer/media_stream_api.cc',
604 'public/renderer/media_stream_api.h', 604 'public/renderer/media_stream_api.h',
605 'public/renderer/media_stream_audio_sink.cc', 605 'public/renderer/media_stream_audio_sink.cc',
606 'public/renderer/media_stream_audio_sink.h', 606 'public/renderer/media_stream_audio_sink.h',
607 'public/renderer/media_stream_renderer_factory.h', 607 'public/renderer/media_stream_renderer_factory.h',
608 'public/renderer/media_stream_sink.h', 608 'public/renderer/media_stream_sink.h',
609 'public/renderer/media_stream_video_sink.cc', 609 'public/renderer/media_stream_video_sink.cc',
610 'public/renderer/media_stream_video_sink.h', 610 'public/renderer/media_stream_video_sink.h',
611 'public/renderer/webrtc_ip_handling_policy.cc',
612 'public/renderer/webrtc_ip_handling_policy.h',
611 'public/renderer/webrtc_log_message_delegate.h', 613 'public/renderer/webrtc_log_message_delegate.h',
612 ], 614 ],
613 # WebRTC-specific sources. Put WebRTC plugin-related stuff in the 615 # WebRTC-specific sources. Put WebRTC plugin-related stuff in the
614 # Plugin+WebRTC section below. 616 # Plugin+WebRTC section below.
615 'private_renderer_webrtc_sources': [ 617 'private_renderer_webrtc_sources': [
616 'renderer/media/media_recorder_handler.cc', 618 'renderer/media/media_recorder_handler.cc',
617 'renderer/media/media_recorder_handler.h', 619 'renderer/media/media_recorder_handler.h',
618 'renderer/media/media_stream.cc', 620 'renderer/media/media_stream.cc',
619 'renderer/media/media_stream.h', 621 'renderer/media/media_stream.h',
620 'renderer/media/media_stream_audio_processor.cc', 622 'renderer/media/media_stream_audio_processor.cc',
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906 }], 908 }],
907 ], 909 ],
908 'target_conditions': [ 910 'target_conditions': [
909 ['OS=="android"', { 911 ['OS=="android"', {
910 'sources/': [ 912 'sources/': [
911 ['include', '^renderer/render_view_linux\\.cc$'], 913 ['include', '^renderer/render_view_linux\\.cc$'],
912 ], 914 ],
913 }], 915 }],
914 ], 916 ],
915 } 917 }
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