From 9087d49b8396ff0f88085f2175aa67d9c0a24c99 Mon Sep 17 00:00:00 2001 From: mbonadei Date: Tue, 25 Apr 2017 00:35:35 -0700 Subject: [PATCH] Enabling 'gn check' on webrtc/video. I disabled the check on "video_tests" because it pulls "//webrtc/media/rtc_unittest_main" as a dependency and it defines the _main (that is already defined by "//webrtc/test:test_main"). I will file a bug to solve this in another CL. BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2832063003 Cr-Commit-Position: refs/heads/master@{#17859} --- .gn | 1 + webrtc/media/BUILD.gn | 6 ++--- webrtc/ortc/BUILD.gn | 3 ++- webrtc/pc/BUILD.gn | 6 +++-- webrtc/video/BUILD.gn | 56 +++++++++++++++++++++++++++++++++++++++++++ 5 files changed, 66 insertions(+), 6 deletions(-) diff --git a/.gn b/.gn index 7d80aa6904..f968276c32 100644 --- a/.gn +++ b/.gn @@ -37,6 +37,7 @@ check_targets = [ "//webrtc/stats/*", "//webrtc/system_wrappers/*", "//webrtc/tools/*", + "//webrtc/video/*", "//webrtc/voice_engine/*", ] diff --git a/webrtc/media/BUILD.gn b/webrtc/media/BUILD.gn index ea4b8f351e..b1f768c52e 100644 --- a/webrtc/media/BUILD.gn +++ b/webrtc/media/BUILD.gn @@ -261,7 +261,7 @@ if (rtc_include_tests) { } } - rtc_source_set("rtc_unittest_main") { + rtc_source_set("rtc_media_tests_utils") { testonly = true include_dirs = [] @@ -313,7 +313,6 @@ if (rtc_include_tests) { "../api/video_codecs:video_codecs_api", "../base:rtc_base", "../base:rtc_base_approved", - "../base:rtc_base_tests_main", "../base:rtc_base_tests_utils", "../call:call_interfaces", "../test:test_support", @@ -441,13 +440,14 @@ if (rtc_include_tests) { deps += [ ":rtc_media", ":rtc_media_base", - ":rtc_unittest_main", + ":rtc_media_tests_utils", "../api:video_frame_api", "../api/audio_codecs:builtin_audio_decoder_factory", "../api/video_codecs:video_codecs_api", "../audio", "../base:rtc_base", "../base:rtc_base_approved", + "../base:rtc_base_tests_main", "../base:rtc_base_tests_utils", "../call:call_interfaces", "../common_video:common_video", diff --git a/webrtc/ortc/BUILD.gn b/webrtc/ortc/BUILD.gn index 1034da632d..3fb105884e 100644 --- a/webrtc/ortc/BUILD.gn +++ b/webrtc/ortc/BUILD.gn @@ -76,8 +76,9 @@ if (rtc_include_tests) { ":ortc", "../base:rtc_base", "../base:rtc_base_approved", + "../base:rtc_base_tests_main", "../base:rtc_base_tests_utils", - "../media:rtc_unittest_main", + "../media:rtc_media_tests_utils", "../p2p:p2p_test_utils", "../p2p:rtc_p2p", "../pc:pc_test_utils", diff --git a/webrtc/pc/BUILD.gn b/webrtc/pc/BUILD.gn index ff99c19097..8e78cafdab 100644 --- a/webrtc/pc/BUILD.gn +++ b/webrtc/pc/BUILD.gn @@ -210,8 +210,9 @@ if (rtc_include_tests) { deps = [ ":libjingle_peerconnection", ":rtc_pc", + "../base:rtc_base_tests_main", "../base:rtc_base_tests_utils", - "../media:rtc_unittest_main", + "../media:rtc_media_tests_utils", "../system_wrappers:metrics_default", ] @@ -356,8 +357,9 @@ if (rtc_include_tests) { ":pc_test_utils", "..:webrtc_common", "../api:fakemetricsobserver", + "../base:rtc_base_tests_main", "../base:rtc_base_tests_utils", - "../media:rtc_unittest_main", + "../media:rtc_media_tests_utils", "../pc:rtc_pc", "../system_wrappers:metrics_default", "//testing/gmock", diff --git a/webrtc/video/BUILD.gn b/webrtc/video/BUILD.gn index 524f84b368..52afc93064 100644 --- a/webrtc/video/BUILD.gn +++ b/webrtc/video/BUILD.gn @@ -60,6 +60,7 @@ rtc_static_library("video") { "../base:rtc_base_approved", "../base:rtc_numerics", "../base:rtc_task_queue", + "../call:call_interfaces", "../common_video", "../logging:rtc_event_log_api", "../media:rtc_media_base", @@ -70,6 +71,8 @@ rtc_static_library("video") { "../modules/rtp_rtcp", "../modules/utility", "../modules/video_coding", + "../modules/video_coding:video_coding_utility", + "../modules/video_coding:webrtc_vp8", "../modules/video_processing", "../system_wrappers", "../voice_engine", @@ -86,8 +89,19 @@ if (rtc_include_tests) { deps = [ "../base:rtc_base_tests_utils", "../base:rtc_task_queue", + "../call:call_interfaces", + "../common_video", + "../logging:rtc_event_log_api", "../media:rtc_media_base", + "../modules/audio_mixer:audio_mixer_impl", + "../modules/rtp_rtcp", + "../modules/video_coding:webrtc_h264", + "../modules/video_coding:webrtc_vp8", + "../modules/video_coding:webrtc_vp9", "../system_wrappers", + "../test:test_common", + "../test:test_support", + "../voice_engine", "//testing/gtest", "//webrtc/test:test_renderer", "//webrtc/test:video_test_common", @@ -105,6 +119,8 @@ if (rtc_include_tests) { ] deps = [ ":video_quality_test", + "../test:field_trial", + "../test:test_support", "//testing/gtest", "//webrtc/test:test_common", ] @@ -129,6 +145,7 @@ if (rtc_include_tests) { "../test:run_test", "../test:test_common", "../test:test_renderer", + "../test:test_support", "//testing/gmock", "//testing/gtest", "//third_party/gflags", @@ -147,11 +164,13 @@ if (rtc_include_tests) { deps = [ ":video_quality_test", + "../base:rtc_base_approved", "../system_wrappers:metrics_default", "../test:field_trial", "../test:run_test", "../test:test_common", "../test:test_renderer", + "../test:test_support", "//third_party/gflags", ] if (!build_with_chromium && is_clang) { @@ -167,12 +186,22 @@ if (rtc_include_tests) { "replay.cc", ] deps = [ + "..:webrtc_common", "../api/video_codecs:video_codecs_api", + "../base:rtc_base_approved", + "../call:call_interfaces", + "../common_video", + "../logging:rtc_event_log_api", + "../modules/rtp_rtcp", + "../system_wrappers", "../system_wrappers:metrics_default", "../test:field_trial", + "../test:rtp_test_utils", "../test:run_test", "../test:test_common", "../test:test_renderer", + "../test:test_support", + "../test:video_test_common", "//third_party/gflags", ] if (!build_with_chromium && is_clang) { @@ -205,7 +234,34 @@ if (rtc_include_tests) { ] deps = [ ":video", + "../api:video_frame_api", + "../api/video_codecs:video_codecs_api", + "../base:rtc_base_approved", + "../base:rtc_base_tests_utils", + "../call:call_interfaces", + "../common_video", + "../logging:rtc_event_log_api", "../media:rtc_media_base", + "../media:rtc_media_tests_utils", + "../modules/pacing", + "../modules/rtp_rtcp", + "../modules/rtp_rtcp:rtp_rtcp_unittests", + "../modules/utility", + "../modules/video_coding", + "../modules/video_coding:video_coding_utility", + "../modules/video_coding:webrtc_h264", + "../modules/video_coding:webrtc_vp8", + "../modules/video_coding:webrtc_vp9", + "../system_wrappers", + "../system_wrappers:field_trial_default", + "../system_wrappers:metrics_api", + "../system_wrappers:metrics_default", + "../test:direct_transport", + "../test:field_trial", + "../test:rtp_test_utils", + "../test:test_common", + "../test:test_support", + "../test:video_test_common", "//testing/gmock", "//testing/gtest", ]