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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 'includes': [ | 6 'includes': [ |
7 '../../build/win_precompile.gypi', | 7 '../../build/win_precompile.gypi', |
8 ], | 8 ], |
9 'target_defaults': { | 9 'target_defaults': { |
10 'defines': [ | 10 'defines': [ |
11 'FEATURE_ENABLE_SSL', | 11 'FEATURE_ENABLE_SSL', |
12 'FEATURE_ENABLE_VOICEMAIL', # TODO(ncarter): Do we really need this? | 12 'FEATURE_ENABLE_VOICEMAIL', # TODO(ncarter): Do we really need this? |
13 '_USE_32BIT_TIME_T', | 13 '_USE_32BIT_TIME_T', |
14 'LOGGING_INSIDE_LIBJINGLE', | 14 'LOGGING_INSIDE_LIBJINGLE', |
15 'EXPAT_RELATIVE_PATH', | 15 'EXPAT_RELATIVE_PATH', |
16 'JSONCPP_RELATIVE_PATH', | 16 'JSONCPP_RELATIVE_PATH', |
17 'WEBRTC_RELATIVE_PATH', | 17 'WEBRTC_RELATIVE_PATH', |
18 'HAVE_WEBRTC_VIDEO', | 18 'HAVE_WEBRTC_VIDEO', |
19 'HAVE_WEBRTC_VOICE', | 19 'HAVE_WEBRTC_VOICE', |
20 'NO_SOUND_SYSTEM', | 20 'NO_SOUND_SYSTEM', |
| 21 'HAVE_SRTP', |
| 22 'SRTP_RELATIVE_PATH', |
21 ], | 23 ], |
22 'configurations': { | 24 'configurations': { |
23 'Debug': { | 25 'Debug': { |
24 'defines': [ | 26 'defines': [ |
25 # TODO(sergeyu): Fix libjingle to use NDEBUG instead of | 27 # TODO(sergeyu): Fix libjingle to use NDEBUG instead of |
26 # _DEBUG and remove this define. See below as well. | 28 # _DEBUG and remove this define. See below as well. |
27 '_DEBUG', | 29 '_DEBUG', |
28 ], | 30 ], |
29 } | 31 } |
30 }, | 32 }, |
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553 'source/talk/session/phone/webrtcvideoframe.h', | 555 'source/talk/session/phone/webrtcvideoframe.h', |
554 'source/talk/session/phone/webrtcvie.h', | 556 'source/talk/session/phone/webrtcvie.h', |
555 'source/talk/session/phone/webrtcvoe.h', | 557 'source/talk/session/phone/webrtcvoe.h', |
556 'source/talk/session/phone/webrtcvoiceengine.cc', | 558 'source/talk/session/phone/webrtcvoiceengine.cc', |
557 'source/talk/session/phone/webrtcvoiceengine.h', | 559 'source/talk/session/phone/webrtcvoiceengine.h', |
558 ], | 560 ], |
559 'conditions': [ | 561 'conditions': [ |
560 ['OS!="android"', { | 562 ['OS!="android"', { |
561 'dependencies': [ | 563 'dependencies': [ |
562 # We won't build with WebRTC on Android. | 564 # We won't build with WebRTC on Android. |
| 565 '<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp', |
563 '<(DEPTH)/third_party/webrtc/modules/modules.gyp:video_capture_modul
e', | 566 '<(DEPTH)/third_party/webrtc/modules/modules.gyp:video_capture_modul
e', |
564 '<(DEPTH)/third_party/webrtc/modules/modules.gyp:video_render_module
', | 567 '<(DEPTH)/third_party/webrtc/modules/modules.gyp:video_render_module
', |
565 '<(DEPTH)/third_party/webrtc/video_engine/video_engine.gyp:video_eng
ine_core', | 568 '<(DEPTH)/third_party/webrtc/video_engine/video_engine.gyp:video_eng
ine_core', |
566 '<(DEPTH)/third_party/webrtc/voice_engine/voice_engine.gyp:voice_eng
ine_core', | 569 '<(DEPTH)/third_party/webrtc/voice_engine/voice_engine.gyp:voice_eng
ine_core', |
567 '<(DEPTH)/third_party/webrtc/system_wrappers/source/system_wrappers.
gyp:system_wrappers', | 570 '<(DEPTH)/third_party/webrtc/system_wrappers/source/system_wrappers.
gyp:system_wrappers', |
568 'libjingle', | 571 'libjingle', |
569 'libjingle_p2p', | 572 'libjingle_p2p', |
570 ], | 573 ], |
571 }], | 574 }], |
572 ], | 575 ], |
573 }, # target libjingle_peerconnection | 576 }, # target libjingle_peerconnection |
574 ], | 577 ], |
575 } | 578 } |
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