Fix the number of frames used when interleaving in AudioBuffer::InterleaveTo()

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 412 matching lines @@
     }
     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(), proc_num_frames_, num_channels_,
+    Interleave(data_ptr->ibuf()->channels(), output_num_frames_, num_channels_,
                frame->data_);
   } else {
-    UpmixMonoToInterleaved(data_ptr->ibuf()->channels()[0], proc_num_frames_,
+    UpmixMonoToInterleaved(data_ptr->ibuf()->channels()[0], output_num_frames_,
                            frame->num_channels_, frame->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