Add mute state field to AudioFrame.

webrtc/modules/audio_processing/aec_dump/aec_dump_impl.cc

 /*
  *  Copyright (c) 2017 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 112 matching lines @@
 }
 
 void AecDumpImpl::WriteRenderStreamMessage(const AudioFrame& frame) {
   auto task = CreateWriteToFileTask();
   auto* event = task->GetEvent();
 
   event->set_type(audioproc::Event::REVERSE_STREAM);
   audioproc::ReverseStream* msg = event->mutable_reverse_stream();
   const size_t data_size =
       sizeof(int16_t) * frame.samples_per_channel_ * frame.num_channels_;
-  msg->set_data(frame.data_, data_size);
+  msg->set_data(frame.data(), data_size);
 
   worker_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>(std::move(task)));
 }
 
 void AecDumpImpl::WriteRenderStreamMessage(const FloatAudioFrame& src) {
   auto task = CreateWriteToFileTask();
   auto* event = task->GetEvent();
 
   event->set_type(audioproc::Event::REVERSE_STREAM);
 
@@ skipping 55 matching lines @@
   RTC_DCHECK(worker_queue);
   RTC_DCHECK(handle);
   std::unique_ptr<FileWrapper> debug_file(FileWrapper::Create());
   if (!debug_file->OpenFromFileHandle(handle)) {
     return nullptr;
   }
   return rtc::MakeUnique<AecDumpImpl>(std::move(debug_file), max_log_size_bytes,
                                       worker_queue);
 }
 }  // namespace webrtc