Move webrtc/audio_*.h to webrtc/api/call

webrtc/pc/BUILD.gn

 # Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
 #
 # Use of this source code is governed by a BSD-style license
 # that can be found in the LICENSE file in the root of the source
 # tree. An additional intellectual property rights grant can be found
 # in the file PATENTS.  All contributing project authors may
 # be found in the AUTHORS file in the root of the source tree.
 
 import("../build/webrtc.gni")
 import("//testing/test.gni")
@@ skipping 31 matching lines @@
     "mediasession.h",
     "mediasink.h",
     "rtcpmuxfilter.cc",
     "rtcpmuxfilter.h",
     "srtpfilter.cc",
     "srtpfilter.h",
     "voicechannel.h",
   ]
 
   deps = [
+    "../api:call_api",
     "../base:rtc_base",
     "../media",
   ]
 
   if (build_with_chromium) {
     sources += [
       "externalhmac.cc",
       "externalhmac.h",
     ]
   }
@@ skipping 63 matching lines @@
 
     if (rtc_build_libsrtp) {
       deps += [ "//third_party/libsrtp" ]
     }
 
     if (is_android) {
       deps += [ "//testing/android/native_test:native_test_support" ]
     }
   }
 }