Add mute state field to AudioFrame.

webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc

Index: webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc
diff --git a/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc b/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc
index 1e679af914b3664746971037a6a2b7cc7a60f0bc..8f70d5131e038f4e297c2e9ee8410c49c44caaf6 100644
--- a/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc
+++ b/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc
@@ -42,11 +42,15 @@ namespace webrtc {
 uint32_t CalculateEnergy(const AudioFrame& audioFrame)
 {
     uint32_t energy = 0;
-    for(size_t position = 0; position < audioFrame.samples_per_channel_;
-        position++)
+    if (!audioFrame.muted())
     {
-        // TODO(andrew): this can easily overflow.
-        energy += audioFrame.data_[position] * audioFrame.data_[position];
+        const int16_t* frame_data = audioFrame.data();
+        for(size_t position = 0; position < audioFrame.samples_per_channel_;
+            position++)
+        {
+            // TODO(andrew): this can easily overflow.
+            energy += frame_data[position] * frame_data[position];
+        }
     }
     return energy;
 }
@@ -54,24 +58,31 @@ uint32_t CalculateEnergy(const AudioFrame& audioFrame)
 void RampIn(AudioFrame& audioFrame)
 {
     assert(rampSize <= audioFrame.samples_per_channel_);
-    for(size_t i = 0; i < rampSize; i++)
+    if (!audioFrame.muted())
     {
-        audioFrame.data_[i] = static_cast<int16_t>(rampArray[i] *
-                                                   audioFrame.data_[i]);
+        int16_t* frame_data = audioFrame.mutable_data();
+        for(size_t i = 0; i < rampSize; i++)
+        {
+            frame_data[i] = static_cast<int16_t>(rampArray[i] * frame_data[i]);
+        }
     }
 }
 
 void RampOut(AudioFrame& audioFrame)
 {
     assert(rampSize <= audioFrame.samples_per_channel_);
-    for(size_t i = 0; i < rampSize; i++)
+    if (!audioFrame.muted())
     {
-        const size_t rampPos = rampSize - 1 - i;
-        audioFrame.data_[i] = static_cast<int16_t>(rampArray[rampPos] *
-                                                   audioFrame.data_[i]);
+        int16_t* frame_data = audioFrame.mutable_data();
+        for(size_t i = 0; i < rampSize; i++)
+        {
+            const size_t rampPos = rampSize - 1 - i;
+            frame_data[i] = static_cast<int16_t>(rampArray[rampPos] *
+                                                 frame_data[i]);
+        }
+        memset(&frame_data[rampSize], 0,
+               (audioFrame.samples_per_channel_ - rampSize) *
+               sizeof(frame_data[0]));
     }
-    memset(&audioFrame.data_[rampSize], 0,
-           (audioFrame.samples_per_channel_ - rampSize) *
-           sizeof(audioFrame.data_[0]));
 }
 }  // namespace webrtc