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Side by Side Diff: webrtc/modules/audio_coding/BUILD.gn

Issue 2180853006: GN conversion of audio_decoder_unittests (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: GN vesion for audio_decoder_unittests Created 4 years, 4 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 import("../../build/webrtc.gni") 9 import("../../build/webrtc.gni")
10 import("audio_coding.gni") 10 import("audio_coding.gni")
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990 "../../:webrtc_common", 990 "../../:webrtc_common",
991 "../../system_wrappers", 991 "../../system_wrappers",
992 "../../system_wrappers:system_wrappers_default", 992 "../../system_wrappers:system_wrappers_default",
993 "../../test:test_support", 993 "../../test:test_support",
994 "../rtp_rtcp", 994 "../rtp_rtcp",
995 "//testing/gtest", 995 "//testing/gtest",
996 "//third_party/gflags:gflags", 996 "//third_party/gflags:gflags",
997 ] 997 ]
998 } # insert_packet_with_timing 998 } # insert_packet_with_timing
999 999
1000 test("audio_decoder_unittests") {
1001 testonly = true
1002 sources = [
1003 "neteq/audio_decoder_unittest.cc",
1004 ]
1005
1006 configs += [ "../..:common_config" ]
1007 public_configs = [ "../..:common_inherited_config" ]
1008
1009 if (is_clang) {
1010 # Suppress warnings from the Chromium Clang plugins (bugs.webrtc.org/163).
1011 configs -= [ "//build/config/clang:find_bad_constructs" ]
1012 }
1013
1014 deps = []
1015
1016 defines = neteq_defines
1017
1018 deps += audio_coding_deps
1019 deps += [
1020 ":audio_decoder_interface",
1021 ":isac",
1022 ":isac_fix",
1023 ":neteq",
1024 ":neteq_unittest_tools",
1025 "../../common_audio/",
1026 "../../test:test_support_main",
1027 "//testing/gtest",
1028 ]
1029
1030 if (is_android) {
1031 deps += [ "//testing/android/native_test:native_test_native_code" ]
1032 }
1033
1034 if (is_ios) {
kjellander_webrtc 2016/08/04 13:25:50 I think this needs to be if (is_android || is_ios)
1035 data = [
1036 "//resources/audio_coding/testfile32kHz.pcm",
1037 ]
1038 }
1039 } # audio_decoder_unittests
1040
1000 if (rtc_enable_protobuf) { 1041 if (rtc_enable_protobuf) {
1001 proto_library("neteq_unittest_proto") { 1042 proto_library("neteq_unittest_proto") {
1002 sources = [ 1043 sources = [
1003 "neteq/neteq_unittest.proto", 1044 "neteq/neteq_unittest.proto",
1004 ] 1045 ]
1005 proto_out_dir = "webrtc/audio_coding/neteq" 1046 proto_out_dir = "webrtc/audio_coding/neteq"
1006 } 1047 }
1007 1048
1008 source_set("rtc_event_log_source") { 1049 source_set("rtc_event_log_source") {
1009 testonly = true 1050 testonly = true
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1225 "//third_party/gflags:gflags", 1266 "//third_party/gflags:gflags",
1226 ] 1267 ]
1227 1268
1228 if (is_clang) { 1269 if (is_clang) {
1229 # Suppress warnings from Chrome's Clang plugins. 1270 # Suppress warnings from Chrome's Clang plugins.
1230 # See http://code.google.com/p/webrtc/issues/detail?id=163 for details. 1271 # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
1231 configs -= [ "//build/config/clang:find_bad_constructs" ] 1272 configs -= [ "//build/config/clang:find_bad_constructs" ]
1232 } 1273 }
1233 } 1274 }
1234 } 1275 }
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