Reformat existing code. There should be no functional effects.

webrtc/modules/audio_device/audio_device_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 388 matching lines @@
     }
 
     _recSamples = nSamples;
     _recSize = _recBytesPerSample*nSamples; // {2,4}*nSamples
     if (_recSize > kMaxBufferSizeBytes)
     {
         assert(false);
         return -1;
     }
 
-    if (nSamples != _recSamples)

[henrika_webrtc, 2015/06/10 08:11:35]

Not sure why these lines are removed. Care to elaborate.

[Peter Kasting, 2015/06/11 04:31:41]

On line 401 above we unconditionally set |_recSamples| equal to |nSamples|. 
Then we make no change to either variable before reaching here.  Therefore, this
conditional can never be true.

-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "invalid number of recorded samples (%d)", nSamples);
-        return -1;
-    }
-
     if (_recChannel == AudioDeviceModule::kChannelBoth)
     {
         // (default) copy the complete input buffer to the local buffer
         memcpy(&_recBuffer[0], audioBuffer, _recSize);
     }
     else
     {
         int16_t* ptr16In = (int16_t*)audioBuffer;
         int16_t* ptr16Out = (int16_t*)&_recBuffer[0];
 
@@ skipping 144 matching lines @@
 // ----------------------------------------------------------------------------
 //  GetPlayoutData
 // ----------------------------------------------------------------------------
 
 int32_t AudioDeviceBuffer::GetPlayoutData(void* audioBuffer)
 {
     CriticalSectionScoped lock(&_critSect);
 
     if (_playSize > kMaxBufferSizeBytes)
     {
-       WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "_playSize %i exceeds "
-       "kMaxBufferSizeBytes in AudioDeviceBuffer::GetPlayoutData", _playSize);
+       WEBRTC_TRACE(kTraceError, kTraceUtility, _id,
+                    "_playSize %i exceeds kMaxBufferSizeBytes in "
+                    "AudioDeviceBuffer::GetPlayoutData", _playSize);
        assert(false);
        return -1;
     }
 
     memcpy(audioBuffer, &_playBuffer[0], _playSize);
 
     if (_playFile.Open())
     {
         // write to binary file in mono or stereo (interleaved)
         _playFile.Write(&_playBuffer[0], _playSize);
     }
 
     return _playSamples;
 }
 
 }  // namespace webrtc