Moved call.h and most of api/call/* into call/

webrtc/api/BUILD.gn

 # Copyright (c) 2015 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")
 if (is_android) {
   import("//build/config/android/config.gni")
   import("//build/config/android/rules.gni")
 }
 
 group("api") {
   public_deps = [
     ":libjingle_peerconnection",
   ]
 }
 
 rtc_source_set("call_api") {
   sources = [
-    "call/audio_receive_stream.h",
-    "call/audio_send_stream.cc",
-    "call/audio_send_stream.h",
     "call/audio_sink.h",
-    "call/audio_state.h",
     "call/flexfec_receive_stream.h",
   ]
 
   deps = [
     # TODO(kjellander): Add remaining dependencies when webrtc:4243 is done.
     ":audio_mixer_api",
     ":transport_api",
     "..:webrtc_common",
     "../base:rtc_base_approved",
     "../modules/audio_coding:audio_encoder_interface",
@@ skipping 293 matching lines @@
 
     public_deps = [
       ":audio_mixer_api",
     ]
 
     deps = [
       "//testing/gmock",
     ]
   }
 }