diff --git a/webrtc/modules/audio_coding/neteq4/audio_multi_vector.cc b/webrtc/modules/audio_coding/neteq4/audio_multi_vector.cc
index baa912c860..b49f8b0e8a 100644
--- a/webrtc/modules/audio_coding/neteq4/audio_multi_vector.cc
+++ b/webrtc/modules/audio_coding/neteq4/audio_multi_vector.cc
@@ -74,8 +74,7 @@ void AudioMultiVector::PushBackInterleaved(const int16_t* append_this,
     return;
   }
   size_t length_per_channel = length / num_channels_;
-  int16_t* temp_array =
-      new int16_t[length_per_channel];  // Intermediate storage.
+  int16_t* temp_array = new int16_t[length_per_channel];  // Temporary storage.
   for (size_t channel = 0; channel < num_channels_; ++channel) {
     // Copy elements to |temp_array|.
     // Set |source_ptr| to first element of this channel.
diff --git a/webrtc/modules/audio_coding/neteq4/neteq_impl.cc b/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
index fb27af2cff..cd69fc9e3b 100644
--- a/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
@@ -143,12 +143,12 @@ int NetEqImpl::InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
   int error = InsertPacketInternal(
       rtp_header, kSyncPayload, sizeof(kSyncPayload), receive_timestamp, true);
 
-   if (error != 0) {
-     LOG_FERR1(LS_WARNING, InsertPacketInternal, error);
-     error_code_ = error;
-     return kFail;
-   }
-   return kOK;
+  if (error != 0) {
+    LOG_FERR1(LS_WARNING, InsertPacketInternal, error);
+    error_code_ = error;
+    return kFail;
+  }
+  return kOK;
 }
 
 int NetEqImpl::GetAudio(size_t max_length, int16_t* output_audio,
diff --git a/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc
index 965f75f2cd..bf5ca7ba5c 100644
--- a/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc
@@ -999,7 +999,7 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(SyncPacketDecode)) {
     // Even if there is RTP packet in NetEq's buffer, the first frame pulled
     // from NetEq starts with few zero samples. Here we measure this delay.
     if (n == 0) {
-      while(decoded[delay_samples] == 0) delay_samples++;
+      while (decoded[delay_samples] == 0) delay_samples++;
     }
     rtp_info.header.sequenceNumber++;
     rtp_info.header.timestamp += kBlockSize16kHz;
@@ -1182,7 +1182,6 @@ void NetEqDecodingTest::WrapTest(uint16_t start_seq_no,
     // Expect delay (in samples) to be less than 2 packets.
     EXPECT_LE(timestamp - neteq_->PlayoutTimestamp(),
               static_cast<uint32_t>(kSamples * 2));
-
   }
   // Make sure we have actually tested wrap-around.
   ASSERT_EQ(expect_seq_no_wrap, seq_no_wrapped);
@@ -1216,4 +1215,4 @@ TEST_F(NetEqDecodingTest, TimestampAndSequenceNumberWrap) {
   WrapTest(0xFFFF - 10, 0xFFFFFFFF - 5000, drop_seq_numbers, true, true);
 }
 
-}  // namespace
+}  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/neteq4/normal.h b/webrtc/modules/audio_coding/neteq4/normal.h
index fa14685f9b..df283198ff 100644
--- a/webrtc/modules/audio_coding/neteq4/normal.h
+++ b/webrtc/modules/audio_coding/neteq4/normal.h
@@ -65,4 +65,4 @@ class Normal {
 };
 
 }  // namespace webrtc
-#endif  // SRC_MODULES_AUDIO_CODING_NETEQ4_NORMAL_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_NORMAL_H_
diff --git a/webrtc/modules/audio_coding/neteq4/random_vector.cc b/webrtc/modules/audio_coding/neteq4/random_vector.cc
index 823909f135..e7a5a1d1bc 100644
--- a/webrtc/modules/audio_coding/neteq4/random_vector.cc
+++ b/webrtc/modules/audio_coding/neteq4/random_vector.cc
@@ -54,4 +54,4 @@ void RandomVector::IncreaseSeedIncrement(int16_t increase_by) {
   seed_increment_+= increase_by;
   seed_increment_ &= kRandomTableSize - 1;
 }
-}
+}  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/neteq4/rtcp.cc b/webrtc/modules/audio_coding/neteq4/rtcp.cc
index f9dcf44919..bc178fc3aa 100644
--- a/webrtc/modules/audio_coding/neteq4/rtcp.cc
+++ b/webrtc/modules/audio_coding/neteq4/rtcp.cc
@@ -10,9 +10,10 @@
 
 #include "webrtc/modules/audio_coding/neteq4/rtcp.h"
 
-#include <algorithm>
 #include <string.h>
 
+#include <algorithm>
+
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/interface/module_common_types.h"