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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 import("//build/config/crypto.gni") | 5 import("//build/config/crypto.gni") |
| 6 import("//build/config/features.gni") | 6 import("//build/config/features.gni") |
| 7 | 7 |
| 8 # From third_party/libjingle/libjingle.gyp's target_defaults. | 8 # From third_party/libjingle/libjingle.gyp's target_defaults. |
| 9 config("jingle_unexported_configs") { | 9 config("jingle_unexported_configs") { |
| 10 defines = [ | 10 defines = [ |
| 11 "EXPAT_RELATIVE_PATH", | 11 "EXPAT_RELATIVE_PATH", |
| 12 "FEATURE_ENABLE_SSL", | 12 "FEATURE_ENABLE_SSL", |
| 13 "GTEST_RELATIVE_PATH", | 13 "GTEST_RELATIVE_PATH", |
| 14 "HAVE_OPENSSL_SSL_H", | 14 "HAVE_OPENSSL_SSL_H", |
| 15 "HAVE_SRTP", | 15 "HAVE_SRTP", |
| 16 "HAVE_WEBRTC_VIDEO", | 16 "HAVE_WEBRTC_VIDEO", |
| 17 "HAVE_WEBRTC_VOICE", | 17 "HAVE_WEBRTC_VOICE", |
| 18 "LOGGING_INSIDE_WEBRTC", | 18 "LOGGING_INSIDE_WEBRTC", |
| 19 "NO_MAIN_THREAD_WRAPPING", | 19 "NO_MAIN_THREAD_WRAPPING", |
| 20 "NO_SOUND_SYSTEM", | 20 "NO_SOUND_SYSTEM", |
| 21 "SRTP_RELATIVE_PATH", | 21 "SRTP_RELATIVE_PATH", |
| 22 "SSL_USE_OPENSSL", | 22 "SSL_USE_OPENSSL", |
| 23 "USE_WEBRTC_DEV_BRANCH", | 23 "USE_WEBRTC_DEV_BRANCH", |
| 24 "ENABLE_EXTERNAL_AUTH", | 24 "ENABLE_EXTERNAL_AUTH", |
| 25 "WEBRTC_CHROMIUM_BUILD", | 25 "WEBRTC_CHROMIUM_BUILD", |
| 26 ] | 26 ] |
| 27 | 27 |
| 28 include_dirs = [ | 28 include_dirs = [ |
| 29 "overrides", | 29 "overrides", |
| 30 "../../third_party/webrtc/overrides", | 30 "../../third_party/webrtc_overrides", |
| 31 "source", | 31 "source", |
| 32 "../../testing/gtest/include", | 32 "../../testing/gtest/include", |
| 33 "../../third_party", | 33 "../../third_party", |
| 34 "../../third_party/libyuv/include", | 34 "../../third_party/libyuv/include", |
| 35 "../../third_party/usrsctp", | 35 "../../third_party/usrsctp", |
| 36 ] | 36 ] |
| 37 | 37 |
| 38 # Assumes libpeer is linked statically. | 38 # Assumes libpeer is linked statically. |
| 39 defines += [ "LIBPEERCONNECTION_LIB=1" ] | 39 defines += [ "LIBPEERCONNECTION_LIB=1" ] |
| 40 | 40 |
| 41 if (is_win && current_cpu == "x86") { | 41 if (is_win && current_cpu == "x86") { |
| 42 defines += [ "_USE_32BIT_TIME_T" ] | 42 defines += [ "_USE_32BIT_TIME_T" ] |
| 43 } | 43 } |
| 44 } | 44 } |
| 45 | 45 |
| 46 # From third_party/libjingle/libjingle.gyp's target_defaults. | 46 # From third_party/libjingle/libjingle.gyp's target_defaults. |
| 47 config("jingle_public_configs") { | 47 config("jingle_public_configs") { |
| 48 include_dirs = [ | 48 include_dirs = [ |
| 49 "../../third_party/webrtc/overrides", | 49 "../../third_party/webrtc_overrides", |
| 50 "overrides", | 50 "overrides", |
| 51 "source", | 51 "source", |
| 52 "../../testing/gtest/include", | 52 "../../testing/gtest/include", |
| 53 "../../third_party", | 53 "../../third_party", |
| 54 ] | 54 ] |
| 55 defines = [ | 55 defines = [ |
| 56 "FEATURE_ENABLE_SSL", | 56 "FEATURE_ENABLE_SSL", |
| 57 "FEATURE_ENABLE_VOICEMAIL", | 57 "FEATURE_ENABLE_VOICEMAIL", |
| 58 "EXPAT_RELATIVE_PATH", | 58 "EXPAT_RELATIVE_PATH", |
| 59 "GTEST_RELATIVE_PATH", | 59 "GTEST_RELATIVE_PATH", |
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| 503 "$p2p_dir/stunprober/stunprober.cc", | 503 "$p2p_dir/stunprober/stunprober.cc", |
| 504 ] | 504 ] |
| 505 | 505 |
| 506 deps = [ | 506 deps = [ |
| 507 "//third_party/webrtc/base:rtc_base", | 507 "//third_party/webrtc/base:rtc_base", |
| 508 ":libjingle_webrtc_common", | 508 ":libjingle_webrtc_common", |
| 509 ] | 509 ] |
| 510 } | 510 } |
| 511 } # enable_webrtc | 511 } # enable_webrtc |
| 512 # TODO(GYP): Port libjingle.gyp's enable_webrtc condition block. | 512 # TODO(GYP): Port libjingle.gyp's enable_webrtc condition block. |
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