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}
diff --git a/webrtc/media/BUILD.gn b/webrtc/media/BUILD.gn
index ea4b8f3..b1f768c 100644
--- a/webrtc/media/BUILD.gn
+++ b/webrtc/media/BUILD.gn
@@ -261,7 +261,7 @@
}
}
- rtc_source_set("rtc_unittest_main") {
+ rtc_source_set("rtc_media_tests_utils") {
testonly = true
include_dirs = []
@@ -313,7 +313,6 @@
"../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 @@
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 1034da6..3fb1058 100644
--- a/webrtc/ortc/BUILD.gn
+++ b/webrtc/ortc/BUILD.gn
@@ -76,8 +76,9 @@
":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 ff99c19..8e78caf 100644
--- a/webrtc/pc/BUILD.gn
+++ b/webrtc/pc/BUILD.gn
@@ -210,8 +210,9 @@
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 @@
":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 524f84b..52afc93 100644
--- a/webrtc/video/BUILD.gn
+++ b/webrtc/video/BUILD.gn
@@ -60,6 +60,7 @@
"../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 @@
"../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 @@
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 @@
]
deps = [
":video_quality_test",
+ "../test:field_trial",
+ "../test:test_support",
"//testing/gtest",
"//webrtc/test:test_common",
]
@@ -129,6 +145,7 @@
"../test:run_test",
"../test:test_common",
"../test:test_renderer",
+ "../test:test_support",
"//testing/gmock",
"//testing/gtest",
"//third_party/gflags",
@@ -147,11 +164,13 @@
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 @@
"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 @@
]
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",
]
