diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn
index 5147ecc972..e2eeb7f96e 100644
--- a/webrtc/BUILD.gn
+++ b/webrtc/BUILD.gn
@@ -245,15 +245,11 @@ source_set("webrtc") {
 
   deps = [
     ":webrtc_common",
-
-    #"api",  # TODO(kjellander): Enable when bugs.webrtc.org/4256 is fixed.
     "audio",
     "base:rtc_base",
     "call",
     "common_audio",
     "common_video",
-
-    #"media",    # TODO(kjellander): Enable when bugs.webrtc.org/4256 is fixed.
     "modules/audio_coding",
     "modules/audio_conference_mixer",
     "modules/audio_device",
@@ -265,9 +261,6 @@ source_set("webrtc") {
     "modules/utility",
     "modules/video_coding",
     "modules/video_processing",
-
-    #"p2p",  # TODO(kjellander): Enable when bugs.webrtc.org/4256 is fixed.
-    #"pc",  # TODO(kjellander): Enable when bugs.webrtc.org/4256 is fixed.
     "system_wrappers",
     "tools",
     "video",
@@ -276,6 +269,15 @@ source_set("webrtc") {
 
   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) {
diff --git a/webrtc/modules/utility/BUILD.gn b/webrtc/modules/utility/BUILD.gn
index 849ea0239c..5437e4f5f7 100644
--- a/webrtc/modules/utility/BUILD.gn
+++ b/webrtc/modules/utility/BUILD.gn
@@ -39,6 +39,10 @@ source_set("utility") {
     configs -= [ "//build/config/clang:find_bad_constructs" ]
   }
 
+  if (is_ios) {
+    libs = [ "AVFoundation.framework" ]
+  }
+
   deps = [
     "../..:webrtc_common",
     "../../base:rtc_task_queue",
diff --git a/webrtc/test/BUILD.gn b/webrtc/test/BUILD.gn
index 451514089a..2966c8d01c 100644
--- a/webrtc/test/BUILD.gn
+++ b/webrtc/test/BUILD.gn
@@ -386,10 +386,12 @@ source_set("test_common") {
     ":rtp_test_utils",
     ":test_support",
     ":video_test_common",
-    "..:webrtc",
     "..:webrtc_common",
+    "../audio",
     "../base:rtc_base_approved",
+    "../call",
     "../modules/media_file",
+    "../video",
     "//testing/gmock",
     "//testing/gtest",
     "//third_party/gflags",