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diff --git a/chromium/third_party/webrtc/modules/audio_processing/rms_level.h b/chromium/third_party/webrtc/modules/audio_processing/rms_level.h
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+++ b/chromium/third_party/webrtc/modules/audio_processing/rms_level.h
@@ -0,0 +1,57 @@
+/*
+ *  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/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
+// Process(). When enough chunks have been accumulated to form a packet, call
+// RMS() to get the audio level indicator for the RTP header.
+class RMSLevel {
+ public:
+  static const int kMinLevel = 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 Process() to accumulate the level.
+  void Process(const int16_t* data, int length);
+
+  // If all samples with the given |length| have a magnitude of zero, this is
+  // a shortcut to avoid some computation.
+  void ProcessMuted(int length);
+
+  // Computes the RMS level over all data passed to Process() since the last
+  // call to RMS(). The returned value is positive but should be interpreted as
+  // negative as per the RFC. It is constrained to [0, 127].
+  int RMS();
+
+ private:
+  float sum_square_;
+  int sample_count_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
+