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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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