diff --git a/modules/audio_processing/audio_buffer.cc b/modules/audio_processing/audio_buffer.cc
index 024b700..eb9bea5 100644
--- a/modules/audio_processing/audio_buffer.cc
+++ b/modules/audio_processing/audio_buffer.cc
@@ -99,7 +99,6 @@
     num_mixed_low_pass_channels_(0),
     reference_copied_(false),
     activity_(AudioFrame::kVadUnknown),
-    is_muted_(false),
     data_(NULL),
     keyboard_data_(NULL),
     channels_(new ChannelBuffer<int16_t>(proc_samples_per_channel_,
@@ -223,7 +222,6 @@
   num_mixed_low_pass_channels_ = 0;
   reference_copied_ = false;
   activity_ = AudioFrame::kVadUnknown;
-  is_muted_ = false;
 }
 
 const int16_t* AudioBuffer::data(int channel) const {
@@ -307,10 +305,6 @@
   return activity_;
 }
 
-bool AudioBuffer::is_muted() const {
-  return is_muted_;
-}
-
 int AudioBuffer::num_channels() const {
   return num_proc_channels_;
 }
@@ -336,9 +330,6 @@
   assert(frame->samples_per_channel_ ==  proc_samples_per_channel_);
   InitForNewData();
   activity_ = frame->vad_activity_;
-  if (frame->energy_ == 0) {
-    is_muted_ = true;
-  }
 
   if (num_proc_channels_ == 1) {
     // We can get away with a pointer assignment in this case.
diff --git a/modules/audio_processing/audio_buffer.h b/modules/audio_processing/audio_buffer.h
index 2b93510..c05ffc9 100644
--- a/modules/audio_processing/audio_buffer.h
+++ b/modules/audio_processing/audio_buffer.h
@@ -71,8 +71,6 @@
   void set_activity(AudioFrame::VADActivity activity);
   AudioFrame::VADActivity activity() const;
 
-  bool is_muted() const;
-
   // Use for int16 interleaved data.
   void DeinterleaveFrom(AudioFrame* audioFrame);
   void InterleaveTo(AudioFrame* audioFrame) const;
@@ -106,7 +104,6 @@
   int num_mixed_low_pass_channels_;
   bool reference_copied_;
   AudioFrame::VADActivity activity_;
-  bool is_muted_;
 
   // If non-null, use this instead of channels_->channel(0). This is an
   // optimization for the case num_proc_channels_ == 1 that allows us to point
diff --git a/modules/audio_processing/level_estimator_impl.cc b/modules/audio_processing/level_estimator_impl.cc
index d209d40..cfe295a 100644
--- a/modules/audio_processing/level_estimator_impl.cc
+++ b/modules/audio_processing/level_estimator_impl.cc
@@ -12,6 +12,7 @@
 
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/audio_processing/rms_level.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 
 namespace webrtc {
@@ -29,13 +30,8 @@
   }
 
   RMSLevel* rms_level = static_cast<RMSLevel*>(handle(0));
-  if (audio->is_muted()) {
-    rms_level->ProcessMuted(audio->samples_per_channel() *
-                            audio->num_channels());
-  } else {
-    for (int i = 0; i < audio->num_channels(); ++i) {
-      rms_level->Process(audio->data(i), audio->samples_per_channel());
-    }
+  for (int i = 0; i < audio->num_channels(); ++i) {
+    rms_level->Process(audio->data(i), audio->samples_per_channel());
   }
 
   return AudioProcessing::kNoError;
diff --git a/modules/audio_processing/rms_level.cc b/modules/audio_processing/rms_level.cc
index 963622b..14136bf 100644
--- a/modules/audio_processing/rms_level.cc
+++ b/modules/audio_processing/rms_level.cc
@@ -15,16 +15,16 @@
 
 namespace webrtc {
 
-static const float kMaxSquaredLevel = 32768.0 * 32768.0;
+static const float kMaxSquaredLevel = 32768 * 32768;
 
 RMSLevel::RMSLevel()
-    : sum_square_(0.0),
+    : sum_square_(0),
       sample_count_(0) {}
 
 RMSLevel::~RMSLevel() {}
 
 void RMSLevel::Reset() {
-  sum_square_ = 0.0;
+  sum_square_ = 0;
   sample_count_ = 0;
 }
 
@@ -40,7 +40,7 @@
 }
 
 int RMSLevel::RMS() {
-  if (sample_count_ == 0 || sum_square_ == 0.0) {
+  if (sample_count_ == 0 || sum_square_ == 0) {
     Reset();
     return kMinLevel;
   }
diff --git a/modules/audio_processing/rms_level.h b/modules/audio_processing/rms_level.h
index 1b19803..055d271 100644
--- a/modules/audio_processing/rms_level.h
+++ b/modules/audio_processing/rms_level.h
@@ -8,6 +8,9 @@
  *  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 {
@@ -49,3 +52,6 @@
 };
 
 }  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
+
diff --git a/modules/audio_processing/test/audio_processing_unittest.cc b/modules/audio_processing/test/audio_processing_unittest.cc
index 0c5b67d..bfb7766 100644
--- a/modules/audio_processing/test/audio_processing_unittest.cc
+++ b/modules/audio_processing/test/audio_processing_unittest.cc
@@ -1229,15 +1229,6 @@
   EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
   EXPECT_EQ(70, apm_->level_estimator()->RMS());
 
-  // Min value if energy_ == 0.
-  SetFrameTo(frame_, 10000);
-  uint32_t energy = frame_->energy_;  // Save default to restore below.
-  frame_->energy_ = 0;
-  EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
-  EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
-  EXPECT_EQ(127, apm_->level_estimator()->RMS());
-  frame_->energy_ = energy;
-
   // Verify reset after enable/disable.
   SetFrameTo(frame_, 32767);
   EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
diff --git a/modules/utility/source/audio_frame_operations.cc b/modules/utility/source/audio_frame_operations.cc
index 18dba52..e3b0010 100644
--- a/modules/utility/source/audio_frame_operations.cc
+++ b/modules/utility/source/audio_frame_operations.cc
@@ -72,7 +72,6 @@
 void AudioFrameOperations::Mute(AudioFrame& frame) {
   memset(frame.data_, 0, sizeof(int16_t) *
       frame.samples_per_channel_ * frame.num_channels_);
-  frame.energy_ = 0;
 }
 
 int AudioFrameOperations::Scale(float left, float right, AudioFrame& frame) {
diff --git a/modules/utility/source/audio_frame_operations_unittest.cc b/modules/utility/source/audio_frame_operations_unittest.cc
index 34c08a8..f4d881c 100644
--- a/modules/utility/source/audio_frame_operations_unittest.cc
+++ b/modules/utility/source/audio_frame_operations_unittest.cc
@@ -142,17 +142,13 @@
 
 TEST_F(AudioFrameOperationsTest, MuteSucceeds) {
   SetFrameData(&frame_, 1000, 1000);
-  frame_.energy_ = 1000 * 1000 * frame_.samples_per_channel_ *
-      frame_.num_channels_;
   AudioFrameOperations::Mute(frame_);
 
   AudioFrame muted_frame;
   muted_frame.samples_per_channel_ = 320;
   muted_frame.num_channels_ = 2;
   SetFrameData(&muted_frame, 0, 0);
-  muted_frame.energy_ = 0;
   VerifyFramesAreEqual(muted_frame, frame_);
-  EXPECT_EQ(muted_frame.energy_, frame_.energy_);
 }
 
 // TODO(andrew): should not allow negative scales.
diff --git a/voice_engine/channel.cc b/voice_engine/channel.cc
index 6bbce78..f919c3d 100644
--- a/voice_engine/channel.cc
+++ b/voice_engine/channel.cc
@@ -3689,9 +3689,9 @@
         MixOrReplaceAudioWithFile(mixingFrequency);
     }
 
-    if (Mute())
-    {
-        AudioFrameOperations::Mute(_audioFrame);
+    bool is_muted = Mute();  // Cache locally as Mute() takes a lock.
+    if (is_muted) {
+      AudioFrameOperations::Mute(_audioFrame);
     }
 
     if (channel_state_.Get().input_external_media)
@@ -3714,7 +3714,11 @@
 
     if (_includeAudioLevelIndication) {
       int length = _audioFrame.samples_per_channel_ * _audioFrame.num_channels_;
-      rms_level_.Process(_audioFrame.data_, length);
+      if (is_muted) {
+        rms_level_.ProcessMuted(length);
+      } else {
+        rms_level_.Process(_audioFrame.data_, length);
+      }
     }
 
     return 0;