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Issue 1357913002: Update to use third_party/webrtc_overrides instead of third_party/webrtc/overrides. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@add_webrtc_overrides
Patch Set: Rebase. Created 5 years, 2 months ago
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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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