Make AudioBuffer::InterleaveTo const

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 43 matching lines @@
     proc_num_frames_(process_num_frames),
     num_proc_channels_(num_process_channels),
     output_num_frames_(output_num_frames),
     num_channels_(num_process_channels),
     num_bands_(NumBandsFromSamplesPerChannel(proc_num_frames_)),
     num_split_frames_(rtc::CheckedDivExact(proc_num_frames_, num_bands_)),
     mixed_low_pass_valid_(false),
     reference_copied_(false),
     activity_(AudioFrame::kVadUnknown),
     keyboard_data_(NULL),
-    data_(new IFChannelBuffer(proc_num_frames_, num_proc_channels_)) {
+    data_(new IFChannelBuffer(proc_num_frames_, num_proc_channels_)),
+    output_buffer_(new IFChannelBuffer(output_num_frames_, num_channels_)) {
   RTC_DCHECK_GT(input_num_frames_, 0);
   RTC_DCHECK_GT(proc_num_frames_, 0);
   RTC_DCHECK_GT(output_num_frames_, 0);
   RTC_DCHECK_GT(num_input_channels_, 0);
   RTC_DCHECK_GT(num_proc_channels_, 0);
   RTC_DCHECK_LE(num_proc_channels_, num_input_channels_);
 
   if (input_num_frames_ != proc_num_frames_ ||
       output_num_frames_ != proc_num_frames_) {
     // Create an intermediate buffer for resampling.
@@ skipping 334 matching lines @@
   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_,
                                      data_->fbuf()->channels()[i],
                                      proc_num_frames_);
     }
   }
 }
 
-void AudioBuffer::InterleaveTo(AudioFrame* frame, bool data_changed) {
+void AudioBuffer::InterleaveTo(AudioFrame* frame, bool data_changed) const {
   frame->vad_activity_ = activity_;
   if (!data_changed) {
     return;
   }
 
   RTC_DCHECK(frame->num_channels_ == num_channels_ || num_channels_ == 1);
   RTC_DCHECK_EQ(frame->samples_per_channel_, output_num_frames_);
 
   // Resample if necessary.
   IFChannelBuffer* data_ptr = data_.get();
   if (proc_num_frames_ != output_num_frames_) {
-    if (!output_buffer_) {
-      output_buffer_.reset(
-          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();
   }
 
   if (frame->num_channels_ == num_channels_) {
     Interleave(data_ptr->ibuf()->channels(), output_num_frames_, num_channels_,
@@ skipping 22 matching lines @@
 
 void AudioBuffer::SplitIntoFrequencyBands() {
   splitting_filter_->Analysis(data_.get(), split_data_.get());
 }
 
 void AudioBuffer::MergeFrequencyBands() {
   splitting_filter_->Synthesis(split_data_.get(), data_.get());
 }
 
 }  // namespace webrtc