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

webrtc/api/call/audio_state.h

Index: webrtc/api/call/audio_state.h
diff --git a/webrtc/api/call/audio_state.h b/webrtc/api/call/audio_state.h
deleted file mode 100644
index b8dca3fb4ec67f9144e69ccb8deb08ad7765b427..0000000000000000000000000000000000000000
--- a/webrtc/api/call/audio_state.h
+++ /dev/null
@@ -1,49 +0,0 @@
-/*
- *  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.
- */
-#ifndef WEBRTC_API_CALL_AUDIO_STATE_H_
-#define WEBRTC_API_CALL_AUDIO_STATE_H_
-
-#include "webrtc/api/audio/audio_mixer.h"
-#include "webrtc/base/refcount.h"
-#include "webrtc/base/scoped_ref_ptr.h"
-
-namespace webrtc {
-
-class VoiceEngine;
-
-// WORK IN PROGRESS
-// This class is under development and is not yet intended for for use outside
-// of WebRtc/Libjingle. Please use the VoiceEngine API instead.
-// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
-
-// AudioState holds the state which must be shared between multiple instances of
-// webrtc::Call for audio processing purposes.
-class AudioState : public rtc::RefCountInterface {
- public:
-  struct Config {
-    // VoiceEngine used for audio streams and audio/video synchronization.
-    // AudioState will tickle the VoE refcount to keep it alive for as long as
-    // the AudioState itself.
-    VoiceEngine* voice_engine = nullptr;
-
-    // The audio mixer connected to active receive streams. One per
-    // AudioState.
-    rtc::scoped_refptr<AudioMixer> audio_mixer;
-  };
-
-  // TODO(solenberg): Replace scoped_refptr with shared_ptr once we can use it.
-  static rtc::scoped_refptr<AudioState> Create(
-      const AudioState::Config& config);
-
-  virtual ~AudioState() {}
-};
-}  // namespace webrtc
-
-#endif  // WEBRTC_API_CALL_AUDIO_STATE_H_