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")
 
@@ skipping 30 matching lines @@
     "mediasession.h",
     "mediasink.h",
     "rtcpmuxfilter.cc",
     "rtcpmuxfilter.h",
     "srtpfilter.cc",
     "srtpfilter.h",
     "voicechannel.h",
   ]
 
   deps = [
+    "../api:audio_api",
     "../base:rtc_base",
     "../media",
   ]
 
   if (build_with_chromium) {
     sources += [
       "externalhmac.cc",
       "externalhmac.h",
     ]
   }
   if (rtc_build_libsrtp) {
     deps += [ "//third_party/libsrtp" ]
   }
 
   configs += [ "..:common_config" ]
   public_configs = [
     "..:common_inherited_config",
     ":rtc_pc_config",
   ]
 
   if (is_clang) {
     # Suppress warnings from Chrome's Clang plugins.
     # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
     configs -= [ "//build/config/clang:find_bad_constructs" ]
   }
 }