Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}

call/audio_state.h

@@ -0,0 +1,55 @@
+/*
+ *  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_CALL_AUDIO_STATE_H_
+#define WEBRTC_CALL_AUDIO_STATE_H_
+
+#include "webrtc/api/audio/audio_mixer.h"
+#include "webrtc/rtc_base/refcount.h"
+#include "webrtc/rtc_base/scoped_ref_ptr.h"
+
+namespace webrtc {
+
+class AudioProcessing;
+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;
+
+    // The audio processing module.
+    rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing;
+  };
+
+  virtual AudioProcessing* audio_processing() = 0;
+
+  // 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_CALL_AUDIO_STATE_H_