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Issue 2720663003: Support GCM ciphers even if ENABLE_EXTERNAL_AUTH is defined. (Closed)
Patch Set: Created 3 years, 9 months ago
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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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114 } 114 }
115 115
116 if (!rtc_libvpx_build_vp9) { 116 if (!rtc_libvpx_build_vp9) {
117 defines += [ "RTC_DISABLE_VP9" ] 117 defines += [ "RTC_DISABLE_VP9" ]
118 } 118 }
119 119
120 if (rtc_enable_sctp) { 120 if (rtc_enable_sctp) {
121 defines += [ "HAVE_SCTP" ] 121 defines += [ "HAVE_SCTP" ]
122 } 122 }
123 123
124 if (rtc_enable_external_auth) {
125 defines += [ "ENABLE_EXTERNAL_AUTH" ]
126 }
127
128 if (build_with_chromium) { 124 if (build_with_chromium) {
129 defines += [ 125 defines += [
130 # NOTICE: Since common_inherited_config is used in public_configs for our 126 # NOTICE: Since common_inherited_config is used in public_configs for our
131 # targets, there's no point including the defines in that config here. 127 # targets, there's no point including the defines in that config here.
132 # TODO(kjellander): Cleanup unused ones and move defines closer to the 128 # TODO(kjellander): Cleanup unused ones and move defines closer to the
133 # source when webrtc:4256 is completed. 129 # source when webrtc:4256 is completed.
134 "HAVE_SRTP", 130 "HAVE_SRTP",
135 "HAVE_WEBRTC_VIDEO", 131 "HAVE_WEBRTC_VIDEO",
136 "HAVE_WEBRTC_VOICE", 132 "HAVE_WEBRTC_VOICE",
137 "LOGGING_INSIDE_WEBRTC", 133 "LOGGING_INSIDE_WEBRTC",
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512 ] 508 ]
513 509
514 deps = [ 510 deps = [
515 "//base:base_java_test_support", 511 "//base:base_java_test_support",
516 "//webrtc/examples:AppRTCMobile_javalib", 512 "//webrtc/examples:AppRTCMobile_javalib",
517 "//webrtc/sdk/android:libjingle_peerconnection_java", 513 "//webrtc/sdk/android:libjingle_peerconnection_java",
518 ] 514 ]
519 } 515 }
520 } 516 }
521 } 517 }
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