Index: webrtc/modules/BUILD.gn |
diff --git a/webrtc/modules/BUILD.gn b/webrtc/modules/BUILD.gn |
index 4ffb31051d35cbb26c1787f02612d538617527be..f1cd14b3fab324fdc0d8e7986220cb144683a0bd 100644 |
--- a/webrtc/modules/BUILD.gn |
+++ b/webrtc/modules/BUILD.gn |
@@ -33,6 +33,81 @@ group("modules") { |
} |
if (rtc_include_tests) { |
+ test("modules_tests") { |
+ testonly = true |
+ |
+ configs += [ "..:common_config" ] |
+ public_configs = [ "..:common_inherited_config" ] |
+ |
+ videoprocessor_defines = [] |
ehmaldonado_webrtc
2016/08/03 15:40:44
Is this the right place to put this?
I'd say so, s
kjellander_webrtc
2016/08/03 15:58:18
Yeah, close to where it's used is best IMO
|
+ if (rtc_use_h264) { |
+ videoprocessor_defines += [ "WEBRTC_VIDEOPROCESSOR_H264_TESTS" ] |
+ } |
+ |
+ defines = audio_coding_defines + videoprocessor_defines |
+ |
+ deps = [ |
+ "..:webrtc_common", |
+ "../common_video", |
+ "../modules/audio_coding", |
+ "../modules/rtp_rtcp", |
+ "../modules/utility", |
+ "../modules/video_coding", |
+ "../modules/video_coding:video_codecs_test_framework", |
+ "../system_wrappers", |
+ "../test:test_support", |
+ "../test:test_support_main", |
+ "//testing/gtest", |
+ ] |
+ |
+ sources = [ |
+ "audio_coding/test/APITest.cc", |
+ "audio_coding/test/Channel.cc", |
+ "audio_coding/test/EncodeDecodeTest.cc", |
+ "audio_coding/test/PCMFile.cc", |
+ "audio_coding/test/PacketLossTest.cc", |
+ "audio_coding/test/RTPFile.cc", |
+ "audio_coding/test/TestAllCodecs.cc", |
+ "audio_coding/test/TestRedFec.cc", |
+ "audio_coding/test/TestStereo.cc", |
+ "audio_coding/test/TestVADDTX.cc", |
+ "audio_coding/test/Tester.cc", |
+ "audio_coding/test/TwoWayCommunication.cc", |
+ "audio_coding/test/iSACTest.cc", |
+ "audio_coding/test/opus_test.cc", |
+ "audio_coding/test/target_delay_unittest.cc", |
+ "audio_coding/test/utility.cc", |
+ "rtp_rtcp/test/testFec/test_fec.cc", |
+ "video_coding/codecs/test/videoprocessor_integrationtest.cc", |
+ "video_coding/codecs/vp8/test/vp8_impl_unittest.cc", |
+ ] |
+ |
+ if (is_android) { |
+ deps += [ "//testing/android/native_test:native_test_native_code" ] |
+ } |
+ if (is_android || is_ios) { |
+ data = [ |
+ "//resources/audio_coding/testfile32kHz.pcm", |
+ "//resources/audio_coding/teststereo32kHz.pcm", |
+ "//resources/foreman_cif.yuv", |
+ "//resources/paris_qcif.yuv", |
+ ] |
+ } |
+ |
+ if (is_clang) { |
+ # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
+ configs -= [ "//build/config/clang:find_bad_constructs" ] |
+ } |
+ if (is_win) { |
+ cflags = [ |
+ # TODO(phoglund): get rid of 4373 supression when |
+ # http://code.google.com/p/webrtc/issues/detail?id=261 is solved. |
+ # legacy warning for ignoring const / volatile in signatures. |
+ "/wd4373", |
+ ] |
+ } |
+ } |
+ |
test("modules_unittests") { |
testonly = true |