Reformat existing code. There should be no functional effects.

webrtc/modules/audio_coding/main/acm2/acm_receive_test.h

 /*
  *  Copyright (c) 2014 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 17 matching lines @@
   enum NumOutputChannels {
     kArbitraryChannels = 0,
     kMonoOutput = 1,
     kStereoOutput = 2
   };
 
   AcmReceiveTest(
       PacketSource* packet_source,
       AudioSink* audio_sink,
       int output_freq_hz,
-      NumOutputChannels exptected_output_channels);
+      NumOutputChannels expected_output_channels);
   virtual ~AcmReceiveTest() {}
 
   // Registers the codecs with default parameters from ACM.
   void RegisterDefaultCodecs();
 
   // Registers codecs with payload types matching the pre-encoded NetEq test
   // files.
   void RegisterNetEqTestCodecs();
 
   // Runs the test and returns true if successful.
   void Run();
 
  private:
   SimulatedClock clock_;
   rtc::scoped_ptr<AudioCoding> acm_;
   PacketSource* packet_source_;
   AudioSink* audio_sink_;
   const int output_freq_hz_;
-  NumOutputChannels exptected_output_channels_;
+  NumOutputChannels expected_output_channels_;
 
   DISALLOW_COPY_AND_ASSIGN(AcmReceiveTest);
 };
 
 }  // namespace test
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVE_TEST_H_