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

webrtc/call/call_unittest.cc

 /*
  *  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.
  */
 
 #include <list>
 #include <memory>
 
-#include "webrtc/api/call/audio_state.h"
-#include "webrtc/call.h"
+#include "webrtc/call/audio_state.h"
+#include "webrtc/call/call.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
+#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
-#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/mock_voice_engine.h"
 
 namespace {
 
 struct CallHelper {
   explicit CallHelper(
       rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory = nullptr)
       : voice_engine_(decoder_factory) {
     webrtc::AudioState::Config audio_state_config;
@@ skipping 255 matching lines @@
   stream = call->CreateFlexfecReceiveStream(config);
   EXPECT_NE(stream, nullptr);
   streams.push_back(stream);
 
   for (auto s : streams) {
     call->DestroyFlexfecReceiveStream(s);
   }
 }
 
 }  // namespace webrtc