Add mute state field to AudioFrame.

webrtc/modules/audio_processing/test/audio_processing_simulator.cc

 /*
  *  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.
  */
 
@@ skipping 12 matching lines @@
 
 namespace webrtc {
 namespace test {
 namespace {
 
 void CopyFromAudioFrame(const AudioFrame& src, ChannelBuffer<float>* dest) {
   RTC_CHECK_EQ(src.num_channels_, dest->num_channels());
   RTC_CHECK_EQ(src.samples_per_channel_, dest->num_frames());
   // Copy the data from the input buffer.
   std::vector<float> tmp(src.samples_per_channel_ * src.num_channels_);
-  S16ToFloat(src.data_, tmp.size(), tmp.data());
+  S16ToFloat(src.data(), tmp.size(), tmp.data());
   Deinterleave(tmp.data(), src.samples_per_channel_, src.num_channels_,
                dest->channels());
 }
 
 std::string GetIndexedOutputWavFilename(const std::string& wav_name,
                                         int counter) {
   std::stringstream ss;
   ss << wav_name.substr(0, wav_name.size() - 4) << "_" << counter
      << wav_name.substr(wav_name.size() - 4);
   return ss.str();
@@ skipping 17 matching lines @@
 
 }  // namespace
 
 SimulationSettings::SimulationSettings() = default;
 SimulationSettings::SimulationSettings(const SimulationSettings&) = default;
 SimulationSettings::~SimulationSettings() = default;
 
 void CopyToAudioFrame(const ChannelBuffer<float>& src, AudioFrame* dest) {
   RTC_CHECK_EQ(src.num_channels(), dest->num_channels_);
   RTC_CHECK_EQ(src.num_frames(), dest->samples_per_channel_);
+  int16_t* dest_data = dest->mutable_data();
   for (size_t ch = 0; ch < dest->num_channels_; ++ch) {
     for (size_t sample = 0; sample < dest->samples_per_channel_; ++sample) {
-      dest->data_[sample * dest->num_channels_ + ch] =
+      dest_data[sample * dest->num_channels_ + ch] =
           src.channels()[ch][sample] * 32767;
     }
   }
 }
 
 AudioProcessingSimulator::AudioProcessingSimulator(
     const SimulationSettings& settings)
     : settings_(settings) {
   if (settings_.ed_graph_output_filename &&
       settings_.ed_graph_output_filename->size() > 0) {
@@ skipping 304 matching lines @@
     size_t kMaxFilenameSize = AudioProcessing::kMaxFilenameSize;
     RTC_CHECK_LE(settings_.aec_dump_output_filename->size(), kMaxFilenameSize);
     RTC_CHECK_EQ(AudioProcessing::kNoError,
                  ap_->StartDebugRecording(
                      settings_.aec_dump_output_filename->c_str(), -1));
   }
 }
 
 }  // namespace test
 }  // namespace webrtc