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Side by Side Diff: third_party/libjingle/libjingle.gyp

Issue 9223031: Enable srtp support in libjingle (Closed) Base URL: http://src.chromium.org/svn/trunk/src/
Patch Set: '' Created 8 years, 10 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 '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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