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/rms_level.h

@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2014 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_RMS_LEVEL_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
+
+#include "webrtc/api/array_view.h"
+#include "webrtc/api/optional.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+// Computes the root mean square (RMS) level in dBFs (decibels from digital
+// full-scale) of audio data. The computation follows RFC 6465:
+// https://tools.ietf.org/html/rfc6465
+// with the intent that it can provide the RTP audio level indication.
+//
+// The expected approach is to provide constant-sized chunks of audio to
+// Analyze(). When enough chunks have been accumulated to form a packet, call
+// Average() to get the audio level indicator for the RTP header.
+class RmsLevel {
+ public:
+  struct Levels {
+    int average;
+    int peak;
+  };
+
+  static constexpr int kMinLevelDb = 127;
+
+  RmsLevel();
+  ~RmsLevel();
+
+  // Can be called to reset internal states, but is not required during normal
+  // operation.
+  void Reset();
+
+  // Pass each chunk of audio to Analyze() to accumulate the level.
+  void Analyze(rtc::ArrayView<const int16_t> data);
+
+  // If all samples with the given |length| have a magnitude of zero, this is
+  // a shortcut to avoid some computation.
+  void AnalyzeMuted(size_t length);
+
+  // Computes the RMS level over all data passed to Analyze() since the last
+  // call to Average(). The returned value is positive but should be interpreted
+  // as negative as per the RFC. It is constrained to [0, 127]. Resets the
+  // internal state to start a new measurement period.
+  int Average();
+
+  // Like Average() above, but also returns the RMS peak value. Resets the
+  // internal state to start a new measurement period.
+  Levels AverageAndPeak();
+
+ private:
+  // Compares |block_size| with |block_size_|. If they are different, calls
+  // Reset() and stores the new size.
+  void CheckBlockSize(size_t block_size);
+
+  float sum_square_;
+  size_t sample_count_;
+  float max_sum_square_;
+  rtc::Optional<size_t> block_size_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
+