Add mute state field to AudioFrame.

webrtc/modules/audio_processing/audio_buffer.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.
  */
 
@@ skipping 375 matching lines @@
         new IFChannelBuffer(input_num_frames_, num_proc_channels_));
   }
   activity_ = frame->vad_activity_;
 
   int16_t* const* deinterleaved;
   if (input_num_frames_ == proc_num_frames_) {
     deinterleaved = data_->ibuf()->channels();
   } else {
     deinterleaved = input_buffer_->ibuf()->channels();
   }
+  // TODO(yujo): handle muted frames more efficiently.
   if (num_proc_channels_ == 1) {
     // Downmix and deinterleave simultaneously.
-    DownmixInterleavedToMono(frame->data_, input_num_frames_,
+    DownmixInterleavedToMono(frame->data(), input_num_frames_,
                              num_input_channels_, deinterleaved[0]);
   } else {
     RTC_DCHECK_EQ(num_proc_channels_, num_input_channels_);
-    Deinterleave(frame->data_,
+    Deinterleave(frame->data(),
                  input_num_frames_,
                  num_proc_channels_,
                  deinterleaved);
   }
 
   // Resample.
   if (input_num_frames_ != proc_num_frames_) {
     for (size_t i = 0; i < num_proc_channels_; ++i) {
       input_resamplers_[i]->Resample(input_buffer_->fbuf_const()->channels()[i],
                                      input_num_frames_,
@@ skipping 20 matching lines @@
           new IFChannelBuffer(output_num_frames_, num_channels_));
     }
     for (size_t i = 0; i < num_channels_; ++i) {
       output_resamplers_[i]->Resample(
           data_->fbuf()->channels()[i], proc_num_frames_,
           output_buffer_->fbuf()->channels()[i], output_num_frames_);
     }
     data_ptr = output_buffer_.get();
   }
 
+  // TODO(yujo): handle muted frames more efficiently.
   if (frame->num_channels_ == num_channels_) {
     Interleave(data_ptr->ibuf()->channels(), output_num_frames_, num_channels_,
-               frame->data_);
+               frame->mutable_data());
   } else {
     UpmixMonoToInterleaved(data_ptr->ibuf()->channels()[0], output_num_frames_,
-                           frame->num_channels_, frame->data_);
+                           frame->num_channels_, frame->mutable_data());
   }
 }
 
 void AudioBuffer::CopyLowPassToReference() {
   reference_copied_ = true;
   if (!low_pass_reference_channels_.get() ||
       low_pass_reference_channels_->num_channels() != num_channels_) {
     low_pass_reference_channels_.reset(
         new ChannelBuffer<int16_t>(num_split_frames_,
                                    num_proc_channels_));
   }
   for (size_t i = 0; i < num_proc_channels_; i++) {
     memcpy(low_pass_reference_channels_->channels()[i],
            split_bands_const(i)[kBand0To8kHz],
            low_pass_reference_channels_->num_frames_per_band() *
                sizeof(split_bands_const(i)[kBand0To8kHz][0]));
   }
 }
 
 void AudioBuffer::SplitIntoFrequencyBands() {
   splitting_filter_->Analysis(data_.get(), split_data_.get());
 }
 
 void AudioBuffer::MergeFrequencyBands() {
   splitting_filter_->Synthesis(split_data_.get(), data_.get());
 }
 
 }  // namespace webrtc