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| 1 # Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 1 # Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. |
| 2 # | 2 # |
| 3 # Use of this source code is governed by a BSD-style license | 3 # Use of this source code is governed by a BSD-style license |
| 4 # that can be found in the LICENSE file in the root of the source | 4 # that can be found in the LICENSE file in the root of the source |
| 5 # tree. An additional intellectual property rights grant can be found | 5 # tree. An additional intellectual property rights grant can be found |
| 6 # in the file PATENTS. All contributing project authors may | 6 # in the file PATENTS. All contributing project authors may |
| 7 # be found in the AUTHORS file in the root of the source tree. | 7 # be found in the AUTHORS file in the root of the source tree. |
| 8 | 8 |
| 9 # TODO(kjellander): Rebase this to webrtc/build/common.gypi changes after r6330. | 9 # TODO(kjellander): Rebase this to webrtc/build/common.gypi changes after r6330. |
| 10 | 10 |
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| 115 } | 115 } |
| 116 | 116 |
| 117 if (!rtc_libvpx_build_vp9) { | 117 if (!rtc_libvpx_build_vp9) { |
| 118 defines += [ "RTC_DISABLE_VP9" ] | 118 defines += [ "RTC_DISABLE_VP9" ] |
| 119 } | 119 } |
| 120 | 120 |
| 121 if (rtc_enable_sctp) { | 121 if (rtc_enable_sctp) { |
| 122 defines += [ "HAVE_SCTP" ] | 122 defines += [ "HAVE_SCTP" ] |
| 123 } | 123 } |
| 124 | 124 |
| 125 if (rtc_enable_external_auth) { |
| 126 defines += [ "ENABLE_EXTERNAL_AUTH" ] |
| 127 } |
| 128 |
| 125 if (build_with_chromium) { | 129 if (build_with_chromium) { |
| 126 defines += [ | 130 defines += [ |
| 127 # NOTICE: Since common_inherited_config is used in public_configs for our | 131 # NOTICE: Since common_inherited_config is used in public_configs for our |
| 128 # targets, there's no point including the defines in that config here. | 132 # targets, there's no point including the defines in that config here. |
| 129 # TODO(kjellander): Cleanup unused ones and move defines closer to the | 133 # TODO(kjellander): Cleanup unused ones and move defines closer to the |
| 130 # source when webrtc:4256 is completed. | 134 # source when webrtc:4256 is completed. |
| 131 "ENABLE_EXTERNAL_AUTH", | |
| 132 "HAVE_OPENSSL_SSL_H", | 135 "HAVE_OPENSSL_SSL_H", |
| 133 "HAVE_SRTP", | 136 "HAVE_SRTP", |
| 134 "HAVE_WEBRTC_VIDEO", | 137 "HAVE_WEBRTC_VIDEO", |
| 135 "HAVE_WEBRTC_VOICE", | 138 "HAVE_WEBRTC_VOICE", |
| 136 "LOGGING_INSIDE_WEBRTC", | 139 "LOGGING_INSIDE_WEBRTC", |
| 137 "SSL_USE_OPENSSL", | 140 "SSL_USE_OPENSSL", |
| 138 "USE_WEBRTC_DEV_BRANCH", | 141 "USE_WEBRTC_DEV_BRANCH", |
| 139 ] | 142 ] |
| 140 } else { | 143 } else { |
| 141 if (is_posix) { | 144 if (is_posix) { |
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| 712 ] | 715 ] |
| 713 | 716 |
| 714 deps = [ | 717 deps = [ |
| 715 "//base:base_java_test_support", | 718 "//base:base_java_test_support", |
| 716 "//webrtc/examples:AppRTCMobile_javalib", | 719 "//webrtc/examples:AppRTCMobile_javalib", |
| 717 "//webrtc/sdk/android:libjingle_peerconnection_java", | 720 "//webrtc/sdk/android:libjingle_peerconnection_java", |
| 718 ] | 721 ] |
| 719 } | 722 } |
| 720 } | 723 } |
| 721 } | 724 } |
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