diff --git a/BUILD.gn b/BUILD.gn
index c2b7ba4..b1301b0 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -6,10 +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 file is copied and modified from Chromium (src/BUILD.gn).
-group("root") {
+group("default") {
+  testonly = true
   deps = [
     "//webrtc",
+    "//webrtc:webrtc_tests",
     "//webrtc/examples",
+    "//webrtc/tools",
   ]
 }
diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn
index f06597d..9e420ba 100644
--- a/webrtc/BUILD.gn
+++ b/webrtc/BUILD.gn
@@ -243,8 +243,12 @@
   precompiled_source = "sdk/objc/WebRTC-Prefix.pch"
 }
 
-if (!is_ios || !build_with_chromium) {
+if (!build_with_chromium) {
+  # Target to build all the WebRTC production code.
   rtc_static_library("webrtc") {
+    # Only the root target should depend on this.
+    visibility = [ "//:default" ]
+
     sources = [
       # TODO(kjellander): Remove this whenever possible. GN's static_library
       # target type requires at least one object to avoid errors linking.
@@ -258,37 +262,84 @@
 
     deps = [
       ":webrtc_common",
+      "api",
       "audio",
-      "base:rtc_base",
+      "base",
       "call",
       "common_audio",
       "common_video",
+      "libjingle/xmllite",
+      "libjingle/xmpp",
+      "logging",
+      "media",
       "modules",
+      "modules/video_capture:video_capture_internal_impl",
+      "p2p",
+      "pc",
+      "sdk",
       "stats",
       "system_wrappers",
-      "tools",
       "video",
       "voice_engine",
     ]
 
-    if (build_with_chromium) {
-      deps += [ "modules/video_capture" ]
-    } else {
-      # TODO(kjellander): Enable for Chromium as well when bugs.webrtc.org/4256
-      # is fixed. Right now it's not possible due to circular dependencies.
-      deps += [
-        "api",
-        "media",
-        "p2p",
-        "pc",
-      ]
-    }
-
     if (rtc_enable_protobuf) {
       defines += [ "ENABLE_RTC_EVENT_LOG" ]
       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",
+        ":xmllite_xmpp_unittests",
+        "api:peerconnection_unittests",
+        "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/rtp_rtcp:test_packet_masks_metrics",
+        "modules/video_capture:video_capture_internal_impl",
+        "pc:rtc_pc_unittests",
+        "stats:rtc_stats_unittests",
+        "system_wrappers:system_wrappers_unittests",
+        "test",
+        "video:screenshare_loopback",
+        "video:video_loopback",
+        "video:video_tests",
+        "voice_engine:voe_cmd_test",
+        "voice_engine:voice_engine_unittests",
+      ]
+      if (is_android) {
+        deps += [
+          ":android_junit_tests",
+          "api:libjingle_peerconnection_android_unittest",
+        ]
+      } else {
+        deps += [ "modules/video_capture:video_capture_tests" ]
+      }
+      if (!is_ios) {
+        deps += [
+          "modules/audio_device:audio_device_tests",
+          "voice_engine:voe_auto_test",
+        ]
+      }
+    }
+  }
 }
 
 rtc_static_library("webrtc_common") {
@@ -637,15 +688,6 @@
     }
   }
 
-  rtc_executable("webrtc_tests") {
-    testonly = true
-    deps = [
-      ":webrtc",
-      "modules/video_capture:video_capture_internal_impl",
-      "test",
-    ]
-  }
-
   rtc_test("webrtc_perf_tests") {
     testonly = true
     configs += [ ":rtc_unittests_config" ]
diff --git a/webrtc/api/BUILD.gn b/webrtc/api/BUILD.gn
index 69fd7f4..3a33796 100644
--- a/webrtc/api/BUILD.gn
+++ b/webrtc/api/BUILD.gn
@@ -16,6 +16,12 @@
   public_deps = [
     ":libjingle_peerconnection",
   ]
+  if (is_android && !build_with_chromium) {
+    public_deps += [
+      ":libjingle_peerconnection_java",
+      ":libjingle_peerconnection_so",
+    ]
+  }
 }
 
 rtc_source_set("call_api") {
diff --git a/webrtc/base/BUILD.gn b/webrtc/base/BUILD.gn
index f3a46f6..aed1158 100644
--- a/webrtc/base/BUILD.gn
+++ b/webrtc/base/BUILD.gn
@@ -19,6 +19,17 @@
   import("//build/config/clang/clang.gni")
 }
 
+group("base") {
+  public_deps = [
+    ":rtc_base",
+    ":rtc_base_approved",
+    ":rtc_task_queue",
+  ]
+  if (is_android) {
+    public_deps += [ ":base_java" ]
+  }
+}
+
 config("rtc_base_approved_all_dependent_config") {
   if (is_mac && !build_with_chromium) {
     libs = [ "Foundation.framework" ]  # needed for logging_mac.mm
diff --git a/webrtc/examples/BUILD.gn b/webrtc/examples/BUILD.gn
index e812f9e..64ff05b 100644
--- a/webrtc/examples/BUILD.gn
+++ b/webrtc/examples/BUILD.gn
@@ -20,12 +20,22 @@
 }
 
 group("examples") {
+  # This target shall build all targets in examples.
+  testonly = true
   public_deps = []
 
   if (is_android) {
+    public_deps += [
+      ":AppRTCMobile",
+      ":AppRTCMobileTest",
+    ]
+  }
+
+  if (is_ios || (is_mac && target_cpu != "x86")) {
     public_deps += [ ":AppRTCMobile" ]
   }
-  if (is_linux) {
+
+  if (is_linux || is_win) {
     public_deps += [
       ":peerconnection_client",
       ":peerconnection_server",
diff --git a/webrtc/logging/BUILD.gn b/webrtc/logging/BUILD.gn
index 54daa20..a50d8e4 100644
--- a/webrtc/logging/BUILD.gn
+++ b/webrtc/logging/BUILD.gn
@@ -13,6 +13,15 @@
   import("//build/config/android/rules.gni")
 }
 
+group("logging") {
+  public_deps = [
+    ":rtc_event_log_impl",
+  ]
+  if (rtc_enable_protobuf) {
+    public_deps += [ ":rtc_event_log_parser" ]
+  }
+}
+
 rtc_source_set("rtc_event_log_api") {
   sources = [
     "rtc_event_log/rtc_event_log.h",
diff --git a/webrtc/modules/BUILD.gn b/webrtc/modules/BUILD.gn
index 8f90a00..a7e388e 100644
--- a/webrtc/modules/BUILD.gn
+++ b/webrtc/modules/BUILD.gn
@@ -19,13 +19,17 @@
     "audio_coding",
     "audio_conference_mixer",
     "audio_device",
-    "audio_mixer:audio_mixer_impl",
+    "audio_mixer",
     "audio_processing",
     "bitrate_controller",
+    "congestion_controller",
     "desktop_capture",
     "media_file",
+    "pacing",
+    "remote_bitrate_estimator",
     "rtp_rtcp",
     "utility",
+    "video_capture",
     "video_coding",
     "video_processing",
   ]
@@ -625,7 +629,7 @@
       "../base:rtc_base",  # TODO(kjellander): Cleanup in bugs.webrtc.org/3806.
       "../common_audio",
       "../common_video",
-      "../system_wrappers:system_wrappers",
+      "../system_wrappers",
       "../test:rtp_test_utils",
       "../test:test_common",
       "../test:test_support",
@@ -645,8 +649,7 @@
       "audio_coding:webrtc_opus",
       "audio_conference_mixer",
       "audio_device",
-      "audio_mixer:audio_frame_manipulator",
-      "audio_mixer:audio_mixer_impl",
+      "audio_mixer",
       "audio_processing",
       "audio_processing:audioproc_test_utils",
       "bitrate_controller",
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index 81e0792..bbf4668 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -913,6 +913,42 @@
 }
 
 if (rtc_include_tests) {
+  group("audio_coding_tests") {
+    testonly = true
+    public_deps = [
+      ":RTPchange",
+      ":RTPencode",
+      ":RTPjitter",
+      ":RTPtimeshift",
+      ":acm_receive_test",
+      ":acm_send_test",
+      ":audio_classifier_test",
+      ":audio_codec_speed_tests",
+      ":audio_decoder_unittests",
+      ":audio_decoder_unittests",
+      ":delay_test",
+      ":g711_test",
+      ":g722_test",
+      ":ilbc_test",
+      ":insert_packet_with_timing",
+      ":isac_api_test",
+      ":isac_fix_test",
+      ":isac_switch_samprate_test",
+      ":isac_test",
+      ":neteq_ilbc_quality_test",
+      ":neteq_isac_quality_test",
+      ":neteq_opus_quality_test",
+      ":neteq_pcmu_quality_test",
+      ":neteq_speed_test",
+      ":rtp_analyze",
+      ":rtpcat",
+      ":webrtc_opus_fec_test",
+    ]
+    if (rtc_enable_protobuf) {
+      public_deps += [ ":neteq_rtpplay" ]
+    }
+  }
+
   rtc_source_set("acm_receive_test") {
     testonly = true
     sources = [
@@ -1028,7 +1064,7 @@
       ":isac_fix",
       ":neteq",
       ":neteq_unittest_tools",
-      "../../common_audio/",
+      "../../common_audio",
       "../../test:test_support_main",
       "//testing/gtest",
     ]
@@ -1128,7 +1164,7 @@
       ":webrtc_opus",
       "../../system_wrappers:system_wrappers_default",
       "../../test:test_support_main",
-      "../audio_processing/",
+      "../audio_processing",
       "//testing/gtest",
     ]
   }
diff --git a/webrtc/modules/audio_mixer/BUILD.gn b/webrtc/modules/audio_mixer/BUILD.gn
index 41aaab0..9b34d54 100644
--- a/webrtc/modules/audio_mixer/BUILD.gn
+++ b/webrtc/modules/audio_mixer/BUILD.gn
@@ -8,6 +8,13 @@
 
 import("../../build/webrtc.gni")
 
+group("audio_mixer") {
+  public_deps = [
+    ":audio_frame_manipulator",
+    ":audio_mixer_impl",
+  ]
+}
+
 rtc_static_library("audio_mixer_impl") {
   visibility = [
     "../../audio:audio",
diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn
index 4090f6d..8aded10 100644
--- a/webrtc/modules/audio_processing/BUILD.gn
+++ b/webrtc/modules/audio_processing/BUILD.gn
@@ -315,6 +315,26 @@
 }
 
 if (rtc_include_tests) {
+  group("audio_processing_tests") {
+    testonly = true
+    public_deps = [
+      ":audioproc_f",
+      ":audioproc_test_utils",
+      ":click_annotate",
+      ":nonlinear_beamformer_test",
+      ":transient_suppression_test",
+      ":unpack_aecdump",
+    ]
+
+    if (rtc_enable_intelligibility_enhancer) {
+      public_deps += [ ":intelligibility_proc" ]
+    }
+
+    if (rtc_enable_protobuf) {
+      public_deps += [ ":audioproc_unittest_proto" ]
+    }
+  }
+
   rtc_executable("unpack_aecdump") {
     testonly = true
     sources = [
diff --git a/webrtc/p2p/BUILD.gn b/webrtc/p2p/BUILD.gn
index a0c4ea7..03cbd6b 100644
--- a/webrtc/p2p/BUILD.gn
+++ b/webrtc/p2p/BUILD.gn
@@ -10,8 +10,13 @@
 
 group("p2p") {
   public_deps = [
+    ":libstunprober",
     ":rtc_p2p",
   ]
+  if (!build_with_chromium) {
+    # TODO(kjellander): Move this to examples or tools.
+    public_deps += [ ":stun_prober" ]
+  }
 }
 
 config("rtc_p2p_inherited_config") {
diff --git a/webrtc/sdk/BUILD.gn b/webrtc/sdk/BUILD.gn
index a1bfbda..0779291 100644
--- a/webrtc/sdk/BUILD.gn
+++ b/webrtc/sdk/BUILD.gn
@@ -15,6 +15,14 @@
   import("//build/config/ios/rules.gni")
 }
 
+group("sdk") {
+  if (is_ios || (is_mac && mac_deployment_target == "10.7")) {
+    public_deps = [
+      ":rtc_sdk_framework_objc",
+    ]
+  }
+}
+
 if (is_ios || (is_mac && mac_deployment_target == "10.7")) {
   config("rtc_sdk_common_objc_config") {
     include_dirs = [
diff --git a/webrtc/tools/BUILD.gn b/webrtc/tools/BUILD.gn
index b94a10f..4ae1b38 100644
--- a/webrtc/tools/BUILD.gn
+++ b/webrtc/tools/BUILD.gn
@@ -10,16 +10,32 @@
 import("../build/webrtc.gni")
 
 group("tools") {
-  deps = [
-    ":command_line_parser",
-  ]
+  # This target shall build all targets in tools/.
+  testonly = true
 
-  if (!build_with_chromium) {
-    # TODO(kjellander): Enable these when webrtc:5970 is fixed.
-    deps += [
-      ":frame_analyzer",
-      ":rgba_to_i420_converter",
+  public_deps = [
+    ":command_line_parser",
+    ":frame_analyzer",
+    ":frame_editor",
+    ":psnr_ssim_analyzer",
+    ":rgba_to_i420_converter",
+  ]
+  if (rtc_include_internal_audio_device) {
+    public_deps += [ ":force_mic_volume_max" ]
+  }
+  if (rtc_enable_protobuf) {
+    public_deps += [ ":chart_proto" ]
+  }
+
+  if (rtc_include_tests) {
+    public_deps += [
+      ":activity_metric",
+      ":rtp_analyzer",
+      ":tools_unittests",
     ]
+    if (rtc_enable_protobuf) {
+      public_deps += [ ":event_log_visualizer" ]
+    }
   }
 }
 
@@ -129,7 +145,7 @@
     }
 
     deps = [
-      "../modules/audio_device:audio_device",
+      "../modules/audio_device",
       "../system_wrappers:system_wrappers_default",
       "//build/win:default_exe_manifest",
     ]
@@ -163,8 +179,8 @@
     deps = [
       "../logging:rtc_event_log_impl",
       "../logging:rtc_event_log_parser",
-      "../modules/congestion_controller:congestion_controller",
-      "../modules/rtp_rtcp:rtp_rtcp",
+      "../modules/congestion_controller",
+      "../modules/rtp_rtcp",
       "../system_wrappers:system_wrappers_default",
       "//build/config/sanitizers:deps",
     ]