diff --git a/BUILD.gn b/BUILD.gn
index e590644526..ef72b41495 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 (is_android) {
 }
 
 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 @@ if (!build_with_chromium) {
       "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 @@ if (!build_with_chromium) {
       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") {