Remove webrtc_tests.
This target is just confusing, since it runs essentially random tests
out of the test binaries it's depending on. This CL makes the top
GN file structured like Chromium's is.
A possible secondary function is to act as a link test (maybe to catch
twice-defined symbols etc), but that seems highly speculative.
Bug: None
Change-Id: Ic8a36ffe96408312e602a58f8806b0694fedaad6
Reviewed-on: https://webrtc-review.googlesource.com/27380
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21028}diff --git a/BUILD.gn b/BUILD.gn
index e590644..ef72b41 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -6,6 +6,12 @@
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
+# This is the root build file for GN. GN will start processing by loading this
+# file, and recursively load all dependencies until all dependencies are either
+# resolved or known not to exist (which will cause the build to fail). So if
+# you add a new build file, there must be some path of dependencies from this
+# file to your new one or GN won't know about it.
+
import("//build/config/linux/pkg_config.gni")
import("//build/config/sanitizers/sanitizers.gni")
import("webrtc.gni")
@@ -16,6 +22,8 @@
}
if (!build_with_chromium) {
+ # This target should (transitively) cause everything to be built; if you run
+ # 'ninja default' and then 'ninja all', the second build should do no work.
group("default") {
testonly = true
deps = [
@@ -24,7 +32,46 @@
"rtc_tools",
]
if (rtc_include_tests) {
- deps += [ ":webrtc_tests" ]
+ deps += [
+ ":rtc_unittests",
+ ":video_engine_tests",
+ ":webrtc_nonparallel_tests",
+ ":webrtc_perf_tests",
+ "common_audio:common_audio_unittests",
+ "common_video:common_video_unittests",
+ "media:rtc_media_unittests",
+ "modules:modules_tests",
+ "modules:modules_unittests",
+ "modules/audio_coding:audio_coding_tests",
+ "modules/audio_processing:audio_processing_tests",
+ "modules/remote_bitrate_estimator:bwe_simulations_tests",
+ "modules/rtp_rtcp:test_packet_masks_metrics",
+ "modules/video_capture:video_capture_internal_impl",
+ "ortc:ortc_unittests",
+ "pc:peerconnection_unittests",
+ "pc:rtc_pc_unittests",
+ "rtc_base:rtc_base_tests_utils",
+ "stats:rtc_stats_unittests",
+ "system_wrappers:system_wrappers_unittests",
+ "test",
+ "video:screenshare_loopback",
+ "video:video_loopback",
+ "voice_engine:voice_engine_unittests",
+ ]
+ if (is_android) {
+ deps += [
+ ":android_junit_tests",
+ "sdk/android:libjingle_peerconnection_android_unittest",
+ ]
+ } else {
+ deps += [ "modules/video_capture:video_capture_tests" ]
+ }
+ if (rtc_enable_protobuf) {
+ deps += [
+ "audio:low_bandwidth_audio_test",
+ "logging:rtc_event_log2rtp_dump",
+ ]
+ }
}
}
}
@@ -294,58 +341,6 @@
deps += [ "logging:rtc_event_log_proto" ]
}
}
-
- if (rtc_include_tests) {
- # Target to build all the WebRTC tests (but not examples or tools).
- # Executable in order to get a target that links all WebRTC code.
- rtc_executable("webrtc_tests") {
- testonly = true
-
- # Only the root target should depend on this.
- visibility = [ "//:default" ]
-
- deps = [
- ":rtc_unittests",
- ":video_engine_tests",
- ":webrtc_nonparallel_tests",
- ":webrtc_perf_tests",
- "common_audio:common_audio_unittests",
- "common_video:common_video_unittests",
- "media:rtc_media_unittests",
- "modules:modules_tests",
- "modules:modules_unittests",
- "modules/audio_coding:audio_coding_tests",
- "modules/audio_processing:audio_processing_tests",
- "modules/remote_bitrate_estimator:bwe_simulations_tests",
- "modules/rtp_rtcp:test_packet_masks_metrics",
- "modules/video_capture:video_capture_internal_impl",
- "ortc:ortc_unittests",
- "pc:peerconnection_unittests",
- "pc:rtc_pc_unittests",
- "rtc_base:rtc_base_tests_utils",
- "stats:rtc_stats_unittests",
- "system_wrappers:system_wrappers_unittests",
- "test",
- "video:screenshare_loopback",
- "video:video_loopback",
- "voice_engine:voice_engine_unittests",
- ]
- if (is_android) {
- deps += [
- ":android_junit_tests",
- "sdk/android:libjingle_peerconnection_android_unittest",
- ]
- } else {
- deps += [ "modules/video_capture:video_capture_tests" ]
- }
- if (rtc_enable_protobuf) {
- deps += [
- "audio:low_bandwidth_audio_test",
- "logging:rtc_event_log2rtp_dump",
- ]
- }
- }
- }
}
rtc_static_library("webrtc_common") {