Reformat existing code. There should be no functional effects.

webrtc/modules/audio_coding/neteq/neteq_impl.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 297 matching lines @@
 // Deprecated.
 // TODO(henrik.lundin) Delete.
 NetEqPlayoutMode NetEqImpl::PlayoutMode() const {
   CriticalSectionScoped lock(crit_sect_.get());
   return playout_mode_;
 }
 
 int NetEqImpl::NetworkStatistics(NetEqNetworkStatistics* stats) {
   CriticalSectionScoped lock(crit_sect_.get());
   assert(decoder_database_.get());
-  const int total_samples_in_buffers = packet_buffer_->NumSamplesInBuffer(
-      decoder_database_.get(), decoder_frame_length_) +
-          static_cast<int>(sync_buffer_->FutureLength());
+  const int total_samples_in_buffers =
+      packet_buffer_->NumSamplesInBuffer(decoder_database_.get(),
+                                         decoder_frame_length_) +
+      static_cast<int>(sync_buffer_->FutureLength());
   assert(delay_manager_.get());
   assert(decision_logic_.get());
   stats_.GetNetworkStatistics(fs_hz_, total_samples_in_buffers,
                               decoder_frame_length_, *delay_manager_.get(),
                               *decision_logic_.get(), stats);
   return 0;
 }
 
 void NetEqImpl::WaitingTimes(std::vector<int>* waiting_times) {
   CriticalSectionScoped lock(crit_sect_.get());
@@ skipping 366 matching lines @@
   } else if (delay_manager_->last_pack_cng_or_dtmf() == -1) {
     // This is first "normal" packet after CNG or DTMF.
     // Reset packet time counter and measure time until next packet,
     // but don't update statistics.
     delay_manager_->set_last_pack_cng_or_dtmf(0);
     delay_manager_->ResetPacketIatCount();
   }
   return 0;
 }
 
-int NetEqImpl::GetAudioInternal(size_t max_length, int16_t* output,
-                                int* samples_per_channel, int* num_channels) {
+int NetEqImpl::GetAudioInternal(size_t max_length,
+                                int16_t* output,
+                                int* samples_per_channel,
+                                int* num_channels) {
   PacketList packet_list;
   DtmfEvent dtmf_event;
   Operations operation;
   bool play_dtmf;
   int return_value = GetDecision(&operation, &packet_list, &dtmf_event,
                                  &play_dtmf);
   if (return_value != 0) {
     LOG_FERR1(LS_WARNING, GetDecision, return_value);
     assert(false);
     last_mode_ = kModeError;
@@ skipping 1271 matching lines @@
 
 void NetEqImpl::CreateDecisionLogic() {
   decision_logic_.reset(DecisionLogic::Create(fs_hz_, output_size_samples_,
                                               playout_mode_,
                                               decoder_database_.get(),
                                               *packet_buffer_.get(),
                                               delay_manager_.get(),
                                               buffer_level_filter_.get()));
 }
 }  // namespace webrtc