Add mute state field to AudioFrame.

webrtc/modules/audio_mixer/audio_frame_manipulator.cc

 /*
  *  Copyright (c) 2012 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 "webrtc/audio/utility/audio_frame_operations.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_mixer/audio_frame_manipulator.h"
 #include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
 uint32_t AudioMixerCalculateEnergy(const AudioFrame& audio_frame) {
   uint32_t energy = 0;
-  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];
+  if (!audio_frame.muted()) {

[hlundin-webrtc, 2017/03/17 14:29:39]

Early return 0 when muted.

[yujo, 2017/03/17 23:55:54]

Done.

+    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 += frame_data[position] * frame_data[position];
+    }
   }
   return energy;
 }
 
 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;
   }
 
   size_t samples = audio_frame->samples_per_channel_;
   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;
   }
 }
 
 void RemixFrame(size_t target_number_of_channels, AudioFrame* frame) {
   RTC_DCHECK_GE(target_number_of_channels, 1);
   RTC_DCHECK_LE(target_number_of_channels, 2);
   if (frame->num_channels_ == 1 && target_number_of_channels == 2) {
     AudioFrameOperations::MonoToStereo(frame);
   } else if (frame->num_channels_ == 2 && target_number_of_channels == 1) {
     AudioFrameOperations::StereoToMono(frame);
   }
 }
 }  // namespace webrtc