| 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",
|
|
|