VoE Utility: Fix a naming nit in RemixAndResample

webrtc/voice_engine/utility.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 23 matching lines @@
 }
 
 void RemixAndResample(const int16_t* src_data,
                       size_t samples_per_channel,
                       size_t num_channels,
                       int sample_rate_hz,
                       PushResampler<int16_t>* resampler,
                       AudioFrame* dst_frame) {
   const int16_t* audio_ptr = src_data;
   size_t audio_ptr_num_channels = num_channels;
-  int16_t downsampled_audio[AudioFrame::kMaxDataSizeSamples];
+  int16_t downsmixed_audio[AudioFrame::kMaxDataSizeSamples];

[ossu, 2017/07/07 11:14:42]

Perhaps take a closer look at what you actually renamed to here. :)

 
   // Downmix before resampling.
   if (num_channels > dst_frame->num_channels_) {
     RTC_DCHECK(num_channels == 2 || num_channels == 4)
         << "num_channels: " << num_channels;
     RTC_DCHECK(dst_frame->num_channels_ == 1 || dst_frame->num_channels_ == 2)
         << "dst_frame->num_channels_: " << dst_frame->num_channels_;
 
     AudioFrameOperations::DownmixChannels(
         src_data, num_channels, samples_per_channel, dst_frame->num_channels_,
-        downsampled_audio);
-    audio_ptr = downsampled_audio;
+        downsmixed_audio);
+    audio_ptr = downsmixed_audio;
     audio_ptr_num_channels = dst_frame->num_channels_;
   }
 
   if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_,
                                     audio_ptr_num_channels) == -1) {
     FATAL() << "InitializeIfNeeded failed: sample_rate_hz = " << sample_rate_hz
             << ", dst_frame->sample_rate_hz_ = " << dst_frame->sample_rate_hz_
             << ", audio_ptr_num_channels = " << audio_ptr_num_channels;
   }
 
@@ skipping 47 matching lines @@
     int32_t temp = 0;
     for (size_t i = 0; i < source_len; ++i) {
       temp = source[i] + target[i];
       target[i] = WebRtcSpl_SatW32ToW16(temp);
     }
   }
 }
 
 }  // namespace voe
 }  // namespace webrtc