Remove unused stuff from AudioFrame:

webrtc/modules/utility/source/audio_frame_operations.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 62 matching lines @@
   for (size_t i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
     int16_t temp_data = frame->data_[i];
     frame->data_[i] = frame->data_[i + 1];
     frame->data_[i + 1] = temp_data;
   }
 }
 
 void AudioFrameOperations::Mute(AudioFrame* frame, bool previous_frame_muted,
                                 bool current_frame_muted) {
   RTC_DCHECK(frame);
-  RTC_DCHECK(frame->interleaved_);
   if (!previous_frame_muted && !current_frame_muted) {
     // Not muted, don't touch.
   } else if (previous_frame_muted && current_frame_muted) {
     // Frame fully muted.
     size_t total_samples = frame->samples_per_channel_ * frame->num_channels_;
     RTC_DCHECK_GE(AudioFrame::kMaxDataSizeSamples, total_samples);
     memset(frame->data_, 0, sizeof(frame->data_[0]) * total_samples);
   } else {
     // Limit number of samples to fade, if frame isn't long enough.
     size_t count = kMuteFadeFrames;
@@ skipping 58 matching lines @@
     } else if (temp_data > 32767) {
       frame.data_[i] = 32767;
     } else {
       frame.data_[i] = static_cast<int16_t>(temp_data);
     }
   }
   return 0;
 }
 
 }  // namespace webrtc