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}

modules/audio_processing/level_controller/gain_applier.h

@@ -0,0 +1,42 @@
+/*
+ *  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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_
+
+#include "webrtc/rtc_base/constructormagic.h"
+
+namespace webrtc {
+
+class ApmDataDumper;
+class AudioBuffer;
+
+class GainApplier {
+ public:
+  explicit GainApplier(ApmDataDumper* data_dumper);
+  void Initialize(int sample_rate_hz);
+
+  // Applies the specified gain to the audio frame and returns the resulting
+  // number of saturated sample values.
+  int Process(float new_gain, AudioBuffer* audio);
+
+ private:
+  ApmDataDumper* const data_dumper_;
+  float old_gain_ = 1.f;
+  float gain_increase_step_size_ = 0.f;
+  float gain_normal_decrease_step_size_ = 0.f;
+  float gain_saturated_decrease_step_size_ = 0.f;
+  bool last_frame_was_saturated_;
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainApplier);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_