Revert of Enable GN check for webrtc/examples (patchset #7 id:120001 of https://codereview.webrtc.org/2714343002/ )

Reason for revert:
I wasn't able to resolve it with that CL so I'll have to revert this by now.

Will have another look at this when time permits.

Original issue's description:
> Enable GN check for webrtc/examples
>
> BUG=webrtc:6828
> NOTRY=True
>
> Review-Url: https://codereview.webrtc.org/2714343002
> Cr-Commit-Position: refs/heads/master@{#16987}
> Committed: https://chromium.googlesource.com/external/webrtc/+/81db74a3841b42c0a84c2f35b91eab66fd3f4e79

TBR=perkj@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6828

Review-Url: https://codereview.webrtc.org/2727253003
Cr-Commit-Position: refs/heads/master@{#16991}
diff --git a/.gn b/.gn
index ef7acf1..2070fa8 100644
--- a/.gn
+++ b/.gn
@@ -27,7 +27,6 @@
   "//webrtc/call/*",
   "//webrtc/common_video/*",
   "//webrtc/common_audio/*",
-  "//webrtc/examples/*",
   "//webrtc/modules/audio_coding/*",
   "//webrtc/modules/audio_conference_mixer/*",
   "//webrtc/modules/audio_device/*",
diff --git a/webrtc/examples/BUILD.gn b/webrtc/examples/BUILD.gn
index 74d340f..d575c65 100644
--- a/webrtc/examples/BUILD.gn
+++ b/webrtc/examples/BUILD.gn
@@ -50,7 +50,6 @@
 
 if (is_android) {
   android_apk("AppRTCMobile") {
-    testonly = true
     apk_name = "AppRTCMobile"
     android_manifest = "androidapp/AndroidManifest.xml"
 
@@ -65,7 +64,6 @@
   }
 
   android_library("AppRTCMobile_javalib") {
-    testonly = true
     android_manifest = "androidapp/AndroidManifest.xml"
 
     java_files = [
@@ -103,7 +101,6 @@
   }
 
   android_resources("AppRTCMobile_resources") {
-    testonly = true
     resource_dirs = [ "androidapp/res" ]
     custom_package = "org.appspot.apprtc"
   }
@@ -172,7 +169,6 @@
   }
 
   rtc_static_library("apprtc_common") {
-    testonly = true
     sources = [
       "objc/AppRTCMobile/common/ARDUtilities.h",
       "objc/AppRTCMobile/common/ARDUtilities.m",
@@ -184,9 +180,9 @@
     public_configs = [ ":apprtc_common_config" ]
 
     deps = [
-      "//webrtc/sdk:rtc_sdk_common_objc",
-      "//webrtc/system_wrappers:field_trial_default",
-      "//webrtc/system_wrappers:metrics_default",
+      "../sdk:rtc_sdk_common_objc",
+      "../system_wrappers:field_trial_default",
+      "../system_wrappers:metrics_default",
     ]
   }
 
@@ -203,7 +199,6 @@
   }
 
   rtc_static_library("apprtc_signaling") {
-    testonly = true
     sources = [
       "objc/AppRTCMobile/ARDAppClient+Internal.h",
       "objc/AppRTCMobile/ARDAppClient.h",
@@ -250,14 +245,13 @@
       ":socketrocket",
     ]
     public_deps = [
-      "//webrtc/sdk:rtc_sdk_peerconnection_objc",
+      "../sdk:rtc_sdk_peerconnection_objc",
     ]
     libs = [ "QuartzCore.framework" ]
   }
 
   if (is_ios) {
     rtc_static_library("AppRTCMobile_lib") {
-      testonly = true
       sources = [
         "objc/AppRTCMobile/ios/ARDAppDelegate.m",
         "objc/AppRTCMobile/ios/ARDMainView.h",
@@ -289,12 +283,10 @@
       deps = [
         ":apprtc_common",
         ":apprtc_signaling",
-        "//webrtc/modules/audio_device",
       ]
     }
 
     ios_app_bundle("AppRTCMobile") {
-      testonly = true
       sources = [
         "objc/AppRTCMobile/ios/main.m",
       ]
@@ -354,7 +346,6 @@
 
   if (is_mac) {
     rtc_static_library("AppRTCMobile_lib") {
-      testonly = true
       sources = [
         "objc/AppRTCMobile/mac/APPRTCAppDelegate.h",
         "objc/AppRTCMobile/mac/APPRTCAppDelegate.m",
@@ -362,7 +353,7 @@
         "objc/AppRTCMobile/mac/APPRTCViewController.m",
       ]
       configs += [
-        "//webrtc:common_objc",
+        "..:common_objc",
         "//build/config/compiler:enable_arc",
       ]
       deps = [
@@ -372,7 +363,6 @@
     }
 
     mac_app_bundle("AppRTCMobile") {
-      testonly = true
       output_name = "AppRTCMobile"
 
       sources = [
@@ -413,7 +403,6 @@
   }
 
   rtc_static_library("socketrocket") {
-    testonly = true
     sources = [
       "objc/AppRTCMobile/third_party/SocketRocket/SRWebSocket.h",
       "objc/AppRTCMobile/third_party/SocketRocket/SRWebSocket.m",
@@ -434,7 +423,6 @@
     # TODO(kthelgason): compile xctests on mac when chromium supports it.
     if (is_ios) {
       rtc_source_set("apprtcmobile_test_sources") {
-        testonly = true
         include_dirs = [
           "objc/AppRTCMobile",
           "objc/AppRTCMobile/ios",
@@ -445,9 +433,6 @@
           "objc/AppRTCMobile/tests/ARDSDPUtils_xctest.mm",
           "objc/AppRTCMobile/tests/ARDSettingsModel_xctest.mm",
         ]
-        deps = [
-          "//webrtc/base:rtc_base",
-        ]
         public_deps = [
           ":AppRTCMobile_ios_frameworks",
           ":AppRTCMobile_lib",
@@ -502,7 +487,6 @@
   }
 
   rtc_executable("peerconnection_client") {
-    testonly = true
     sources = [
       "peerconnection/client/conductor.cc",
       "peerconnection/client/conductor.h",
@@ -516,7 +500,6 @@
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
-    deps = []
     if (is_win) {
       sources += [
         "peerconnection/client/flagdefs.h",
@@ -526,8 +509,13 @@
       ]
       cflags = [ "/wd4245" ]
       configs += [ "//build/config/win:windowed" ]
-      deps += [ "//webrtc/media:rtc_media_base" ]
     }
+    deps = [
+      "//third_party/libyuv",
+      "//webrtc/pc:libjingle_peerconnection",
+      "//webrtc/system_wrappers:field_trial_default",
+      "//webrtc/system_wrappers:metrics_default",
+    ]
     if (is_linux) {
       sources += [
         "peerconnection/client/linux/main.cc",
@@ -543,26 +531,12 @@
       deps += [ "//build/config/linux/gtk" ]
     }
     configs += [ ":peerconnection_client_warnings_config" ]
-
-    deps += [
-      "//third_party/libyuv",
-      "//webrtc/api:libjingle_peerconnection_test_api",
-      "//webrtc/api:video_frame_api",
-      "//webrtc/base:rtc_base",
-      "//webrtc/base:rtc_base_approved",
-      "//webrtc/media:rtc_media",
-      "//webrtc/modules/video_capture:video_capture_module",
-      "//webrtc/pc:libjingle_peerconnection",
-      "//webrtc/system_wrappers:field_trial_default",
-      "//webrtc/system_wrappers:metrics_default",
-    ]
     if (rtc_build_json) {
       deps += [ "//third_party/jsoncpp" ]
     }
   }
 
   rtc_executable("peerconnection_server") {
-    testonly = true
     sources = [
       "peerconnection/server/data_socket.cc",
       "peerconnection/server/data_socket.h",
@@ -583,14 +557,11 @@
     }
   }
   rtc_executable("relayserver") {
-    testonly = true
     sources = [
       "relayserver/relayserver_main.cc",
     ]
     deps = [
-      "../base:rtc_base",
       "//webrtc/base:rtc_base_approved",
-      "//webrtc/p2p:rtc_p2p",
       "//webrtc/pc:rtc_pc",
       "//webrtc/system_wrappers:field_trial_default",
       "//webrtc/system_wrappers:metrics_default",
@@ -601,14 +572,11 @@
     }
   }
   rtc_executable("turnserver") {
-    testonly = true
     sources = [
       "turnserver/turnserver_main.cc",
     ]
     deps = [
-      "../base:rtc_base",
       "//webrtc/base:rtc_base_approved",
-      "//webrtc/p2p:rtc_p2p",
       "//webrtc/pc:rtc_pc",
       "//webrtc/system_wrappers:field_trial_default",
       "//webrtc/system_wrappers:metrics_default",
@@ -619,14 +587,11 @@
     }
   }
   rtc_executable("stunserver") {
-    testonly = true
     sources = [
       "stunserver/stunserver_main.cc",
     ]
     deps = [
-      "../base:rtc_base",
       "//webrtc/base:rtc_base_approved",
-      "//webrtc/p2p:rtc_p2p",
       "//webrtc/pc:rtc_pc",
       "//webrtc/system_wrappers:field_trial_default",
       "//webrtc/system_wrappers:metrics_default",
@@ -641,7 +606,6 @@
 if (!build_with_chromium) {
   # Doesn't build within Chrome on Win.
   rtc_executable("stun_prober") {
-    testonly = true
     sources = [
       "stunprober/main.cc",
     ]
@@ -653,8 +617,6 @@
     }
 
     deps = [
-      "../base:rtc_base",
-      "../base:rtc_base_approved",
       "../p2p:libstunprober",
       "../p2p:rtc_p2p",
       "../system_wrappers:field_trial_default",