GN: Enable api,media,pc and p2p for the 'webrtc' target.

These parts were commented out to avoid breaking the Chromium
WebRTC FYI bots. Include them in the WebRTC build to make our bots
build as many as possible of our GN targets.

BUG=webrtc:5949
NOTRY=True
TBR=phoglund@webrtc.org

Review-Url: https://codereview.webrtc.org/2054903002
Cr-Commit-Position: refs/heads/master@{#13097}

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) {

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",

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",