Index: webrtc/video/BUILD.gn |
diff --git a/webrtc/video/BUILD.gn b/webrtc/video/BUILD.gn |
index 52afc93064771e1878502ec9fd4060dbd9899ef1..20b5d109bd04f6c2efc4e82ecbc98e5e38ddc1e1 100644 |
--- a/webrtc/video/BUILD.gn |
+++ b/webrtc/video/BUILD.gn |
@@ -82,6 +82,7 @@ rtc_static_library("video") { |
if (rtc_include_tests) { |
rtc_source_set("video_quality_test") { |
testonly = true |
+ visibility = [ ":*" ] # Only targets in this file can depend on this. |
sources = [ |
"video_quality_test.cc", |
"video_quality_test.h", |
@@ -114,6 +115,13 @@ if (rtc_include_tests) { |
rtc_source_set("video_full_stack_tests") { |
testonly = true |
+ |
+ # Skip restricting visibility on mobile platforms since the tests on those |
+ # gets additional generated targets which would require many lines here to |
+ # cover (which would be confusing to read and hard to maintain). |
+ if (!is_android && !is_ios) { |
+ visibility = [ "//webrtc:webrtc_perf_tests" ] |
+ } |
sources = [ |
"full_stack_tests.cc", |
] |
@@ -213,6 +221,13 @@ if (rtc_include_tests) { |
# TODO(pbos): Rename test suite. |
rtc_source_set("video_tests") { |
testonly = true |
+ |
+ # Skip restricting visibility on mobile platforms since the tests on those |
+ # gets additional generated targets which would require many lines here to |
+ # cover (which would be confusing to read and hard to maintain). |
+ if (!is_android && !is_ios) { |
+ visibility = [ "//webrtc:video_engine_tests" ] |
+ } |
defines = [] |
sources = [ |
"call_stats_unittest.cc", |
@@ -245,7 +260,7 @@ if (rtc_include_tests) { |
"../media:rtc_media_tests_utils", |
"../modules/pacing", |
"../modules/rtp_rtcp", |
- "../modules/rtp_rtcp:rtp_rtcp_unittests", |
+ "../modules/rtp_rtcp:mock_rtp_rtcp", |
"../modules/utility", |
"../modules/video_coding", |
"../modules/video_coding:video_coding_utility", |