Added an empty AudioTransportProxy to AudioState.

webrtc/call/BUILD.gn

Index: webrtc/call/BUILD.gn
diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn
index 98ea9bcae8376fa52b1c4071e305d55214c9978f..0720aca4b64b4a96c280a4660cc62741e9222bf1 100644
--- a/webrtc/call/BUILD.gn
+++ b/webrtc/call/BUILD.gn
@@ -21,9 +21,12 @@ rtc_static_library("call") {
     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
   }
 
+  public_deps = [
+    "../api:call_api",
+  ]
+
   deps = [
     "..:webrtc_common",
-    "../api:call_api",
     "../audio",
     "../base:rtc_task_queue",
     "../logging:rtc_event_log_impl",
@@ -53,5 +56,13 @@ if (rtc_include_tests) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
+    if (is_win) {
+      cflags = [
+        # TODO(phoglund): get rid of 4373 supression when
+        # http://code.google.com/p/webrtc/issues/detail?id=261 is solved.
+        # legacy warning for ignoring const / volatile in signatures.
+        "/wd4373",
+      ]
+    }
   }
 }