diff --git a/webrtc/modules/BUILD.gn b/webrtc/modules/BUILD.gn
index b7db3d019d..1cd97f3b97 100644
--- a/webrtc/modules/BUILD.gn
+++ b/webrtc/modules/BUILD.gn
@@ -609,6 +609,7 @@ if (rtc_include_tests) {
         "audio_processing/low_cut_filter_unittest.cc",
         "audio_processing/noise_suppression_unittest.cc",
         "audio_processing/residual_echo_detector_unittest.cc",
+        "audio_processing/rms_level_unittest.cc",
         "audio_processing/test/bitexactness_tools.cc",
         "audio_processing/test/bitexactness_tools.h",
         "audio_processing/test/debug_dump_replayer.cc",
diff --git a/webrtc/modules/audio_processing/rms_level_unittest.cc b/webrtc/modules/audio_processing/rms_level_unittest.cc
new file mode 100644
index 0000000000..7b15717ca0
--- /dev/null
+++ b/webrtc/modules/audio_processing/rms_level_unittest.cc
@@ -0,0 +1,92 @@
+/*
+ *  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.
+ */
+#include <cmath>
+#include <memory>
+#include <vector>
+
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/mathutils.h"
+#include "webrtc/base/safe_conversions.h"
+#include "webrtc/modules/audio_processing/rms_level.h"
+#include "webrtc/test/gtest.h"
+
+namespace webrtc {
+namespace {
+constexpr int kSampleRateHz = 48000;
+constexpr size_t kBlockSizeSamples = kSampleRateHz / 100;
+
+std::unique_ptr<RMSLevel> RunTest(rtc::ArrayView<const int16_t> input) {
+  std::unique_ptr<RMSLevel> level(new RMSLevel);
+  for (size_t n = 0; n + kBlockSizeSamples <= input.size();
+       n += kBlockSizeSamples) {
+    level->Process(input.subview(n, kBlockSizeSamples).data(),
+                   kBlockSizeSamples);
+  }
+  return level;
+}
+
+std::vector<int16_t> CreateSinusoid(int frequency_hz,
+                                    int amplitude,
+                                    size_t num_samples) {
+  std::vector<int16_t> x(num_samples);
+  for (size_t n = 0; n < num_samples; ++n) {
+    x[n] = rtc::saturated_cast<int16_t>(
+        amplitude * std::sin(2 * M_PI * n * frequency_hz / kSampleRateHz));
+  }
+  return x;
+}
+}  // namespace
+
+TEST(RmsLevelTest, Run1000HzFullScale) {
+  auto x = CreateSinusoid(1000, INT16_MAX, kSampleRateHz);
+  auto level = RunTest(x);
+  EXPECT_EQ(3, level->RMS());  // -3 dBFS
+}
+
+TEST(RmsLevelTest, Run1000HzHalfScale) {
+  auto x = CreateSinusoid(1000, INT16_MAX / 2, kSampleRateHz);
+  auto level = RunTest(x);
+  EXPECT_EQ(9, level->RMS());  // -9 dBFS
+}
+
+TEST(RmsLevelTest, RunZeros) {
+  std::vector<int16_t> x(kSampleRateHz, 0);  // 1 second of pure silence.
+  auto level = RunTest(x);
+  EXPECT_EQ(127, level->RMS());
+}
+
+TEST(RmsLevelTest, NoSamples) {
+  RMSLevel level;
+  EXPECT_EQ(127, level.RMS());  // Return minimum if no samples are given.
+}
+
+TEST(RmsLevelTest, PollTwice) {
+  auto x = CreateSinusoid(1000, INT16_MAX, kSampleRateHz);
+  auto level = RunTest(x);
+  level->RMS();
+  EXPECT_EQ(127, level->RMS());  // Stats should be reset at this point.
+}
+
+TEST(RmsLevelTest, Reset) {
+  auto x = CreateSinusoid(1000, INT16_MAX, kSampleRateHz);
+  auto level = RunTest(x);
+  level->Reset();
+  EXPECT_EQ(127, level->RMS());  // Stats should be reset at this point.
+}
+
+// Inserts 1 second of full-scale sinusoid, followed by 1 second of muted.
+TEST(RmsLevelTest, ProcessMuted) {
+  auto x = CreateSinusoid(1000, INT16_MAX, kSampleRateHz);
+  auto level = RunTest(x);
+  level->ProcessMuted(kSampleRateHz);
+  EXPECT_EQ(6, level->RMS());  // Average RMS halved due to the silence.
+}
+
+}  // namespace webrtc