Reformat existing code. There should be no functional effects.

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

 /*
  *  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.
  */
 
 #include "webrtc/modules/audio_coding/main/acm2/acm_receive_test.h"
 
 #include <assert.h>
 #include <stdio.h>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
 
 namespace webrtc {
 namespace test {
 
 AcmReceiveTest::AcmReceiveTest(PacketSource* packet_source,
                                AudioSink* audio_sink,
                                int output_freq_hz,
-                               NumOutputChannels exptected_output_channels)
+                               NumOutputChannels expected_output_channels)
     : clock_(0),
       packet_source_(packet_source),
       audio_sink_(audio_sink),
       output_freq_hz_(output_freq_hz),
-      exptected_output_channels_(exptected_output_channels) {
+      expected_output_channels_(expected_output_channels) {
   webrtc::AudioCoding::Config config;
   config.clock = &clock_;
   config.playout_frequency_hz = output_freq_hz_;
   acm_.reset(webrtc::AudioCoding::Create(config));
 }
 
 void AcmReceiveTest::RegisterDefaultCodecs() {
   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kOpus, 120));
   ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISAC, 103));
 #ifndef WEBRTC_ANDROID
@@ skipping 44 matching lines @@
 void AcmReceiveTest::Run() {
   for (rtc::scoped_ptr<Packet> packet(packet_source_->NextPacket()); packet;
        packet.reset(packet_source_->NextPacket())) {
     // Pull audio until time to insert packet.
     while (clock_.TimeInMilliseconds() < packet->time_ms()) {
       AudioFrame output_frame;
       EXPECT_TRUE(acm_->Get10MsAudio(&output_frame));
       EXPECT_EQ(output_freq_hz_, output_frame.sample_rate_hz_);
       const int samples_per_block = output_freq_hz_ * 10 / 1000;
       EXPECT_EQ(samples_per_block, output_frame.samples_per_channel_);
-      if (exptected_output_channels_ != kArbitraryChannels) {
+      if (expected_output_channels_ != kArbitraryChannels) {
         if (output_frame.speech_type_ == webrtc::AudioFrame::kPLC) {
           // Don't check number of channels for PLC output, since each test run
           // usually starts with a short period of mono PLC before decoding the
           // first packet.
         } else {
-          EXPECT_EQ(exptected_output_channels_, output_frame.num_channels_);
+          EXPECT_EQ(expected_output_channels_, output_frame.num_channels_);
         }
       }
       ASSERT_TRUE(audio_sink_->WriteAudioFrame(output_frame));
       clock_.AdvanceTimeMilliseconds(10);
     }
 
     // Insert packet after converting from RTPHeader to WebRtcRTPHeader.
     WebRtcRTPHeader header;
     header.header = packet->header();
     header.frameType = kAudioFrameSpeech;
     memset(&header.type.Audio, 0, sizeof(RTPAudioHeader));
     EXPECT_TRUE(acm_->InsertPacket(packet->payload(),
                                    packet->payload_length_bytes(),
                                    header))
         << "Failure when inserting packet:" << std::endl
         << "  PT = " << static_cast<int>(header.header.payloadType) << std::endl
         << "  TS = " << header.header.timestamp << std::endl
         << "  SN = " << header.header.sequenceNumber;
   }
 }
 
 }  // namespace test
 }  // namespace webrtc