Reformat existing code. There should be no functional effects.

webrtc/modules/audio_coding/neteq/packet_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 232 matching lines @@
 
 int PacketBuffer::NumSamplesInBuffer(DecoderDatabase* decoder_database,
                                      int last_decoded_length) const {
   PacketList::const_iterator it;
   int num_samples = 0;
   int last_duration = last_decoded_length;
   for (it = buffer_.begin(); it != buffer_.end(); ++it) {
     Packet* packet = (*it);
     AudioDecoder* decoder =
         decoder_database->GetDecoder(packet->header.payloadType);
-    if (decoder) {
-      int duration;
-      if (packet->sync_packet) {
-        duration = last_duration;
-      } else if (packet->primary) {
-        duration =
-            decoder->PacketDuration(packet->payload, packet->payload_length);
-      } else {
+    if (decoder && !packet->sync_packet) {
+      if (!packet->primary) {
         continue;
       }
+      int duration =
+        decoder->PacketDuration(packet->payload, packet->payload_length);
       if (duration >= 0) {
         last_duration = duration;  // Save the most up-to-date (valid) duration.
       }
     }
     num_samples += last_duration;
   }
   return num_samples;
 }
 
 void PacketBuffer::IncrementWaitingTimes(int inc) {
@@ skipping 19 matching lines @@
     // Continue while the list is not empty.
   }
 }
 
 void PacketBuffer::BufferStat(int* num_packets, int* max_num_packets) const {
   *num_packets = static_cast<int>(buffer_.size());
   *max_num_packets = static_cast<int>(max_number_of_packets_);
 }
 
 }  // namespace webrtc