Add mute state field to AudioFrame.

webrtc/modules/audio_mixer/audio_frame_manipulator.cc

Index: webrtc/modules/audio_mixer/audio_frame_manipulator.cc
diff --git a/webrtc/modules/audio_mixer/audio_frame_manipulator.cc b/webrtc/modules/audio_mixer/audio_frame_manipulator.cc
index 8aa0b5ce02dc35c6bd3a9d4a90de48ca82f015d8..fff79a38cf6b4e68f13e30c43a56e31a49dd1efc 100644
--- a/webrtc/modules/audio_mixer/audio_frame_manipulator.cc
+++ b/webrtc/modules/audio_mixer/audio_frame_manipulator.cc
@@ -16,11 +16,16 @@
 namespace webrtc {
 
 uint32_t AudioMixerCalculateEnergy(const AudioFrame& audio_frame) {
+  if (audio_frame.muted()) {
+    return 0;
+  }
+
   uint32_t energy = 0;
+  const int16_t* frame_data = audio_frame.data();
   for (size_t position = 0; position < audio_frame.samples_per_channel_;
        position++) {
     // TODO(aleloi): This can overflow. Convert to floats.
-    energy += audio_frame.data_[position] * audio_frame.data_[position];
+    energy += frame_data[position] * frame_data[position];
   }
   return energy;
 }
@@ -29,7 +34,7 @@ void Ramp(float start_gain, float target_gain, AudioFrame* audio_frame) {
   RTC_DCHECK(audio_frame);
   RTC_DCHECK_GE(start_gain, 0.0f);
   RTC_DCHECK_GE(target_gain, 0.0f);
-  if (start_gain == target_gain) {
+  if (start_gain == target_gain || audio_frame->muted()) {
     return;
   }
 
@@ -37,11 +42,12 @@ void Ramp(float start_gain, float target_gain, AudioFrame* audio_frame) {
   RTC_DCHECK_LT(0, samples);
   float increment = (target_gain - start_gain) / samples;
   float gain = start_gain;
+  int16_t* frame_data = audio_frame->mutable_data();
   for (size_t i = 0; i < samples; ++i) {
     // If the audio is interleaved of several channels, we want to
     // apply the same gain change to the ith sample of every channel.
     for (size_t ch = 0; ch < audio_frame->num_channels_; ++ch) {
-      audio_frame->data_[audio_frame->num_channels_ * i + ch] *= gain;
+      frame_data[audio_frame->num_channels_ * i + ch] *= gain;
     }
     gain += increment;
   }