diff --git a/webrtc/modules/audio_coding/neteq4/expand.cc b/webrtc/modules/audio_coding/neteq4/expand.cc
index b9157c363b..73f2ef85a5 100644
--- a/webrtc/modules/audio_coding/neteq4/expand.cc
+++ b/webrtc/modules/audio_coding/neteq4/expand.cc
@@ -415,9 +415,11 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
 
   // Calculate correlation in downsampled domain (4 kHz sample rate).
   int16_t correlation_scale;
-  int correlation_length = Correlation(audio_history, signal_length,
-                                       correlation_vector, &correlation_scale);
-  correlation_length = 51;  // TODO(hlundin): Legacy bit-exactness.
+  int correlation_length = 51;  // TODO(hlundin): Legacy bit-exactness.
+  // If it is decided to break bit-exactness |correlation_length| should be
+  // initialized to the return value of Correlation().
+  Correlation(audio_history, signal_length, correlation_vector,
+              &correlation_scale);
 
   // Find peaks in correlation vector.
   DspHelper::PeakDetection(correlation_vector, correlation_length,
@@ -449,7 +451,7 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
 
   // Find the maximizing index |i| of the cost function
   // f[i] = best_correlation[i] / best_distortion[i].
-  int32_t best_ratio = -1;
+  int32_t best_ratio = std::numeric_limits<int32_t>::min();
   int best_index = -1;
   for (int i = 0; i < kNumCorrelationCandidates; ++i) {
     int32_t ratio;
diff --git a/webrtc/modules/audio_coding/neteq4/neteq_impl.cc b/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
index a5c45ff7b4..73ca5e4e21 100644
--- a/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
@@ -1259,6 +1259,7 @@ int NetEqImpl::DecodeLoop(PacketList* packet_list, Operations* operation,
 
     delete[] packet->payload;
     delete packet;
+    packet = NULL;
     if (decode_length > 0) {
       *decoded_length += decode_length;
       // Update |decoder_frame_length_| with number of samples per channel.
@@ -1287,10 +1288,10 @@ int NetEqImpl::DecodeLoop(PacketList* packet_list, Operations* operation,
     }
   }  // End of decode loop.
 
-  // If the list is not empty at this point, it must hold exactly one CNG
-  // packet.
-  assert(packet_list->empty() ||
-         (packet_list->size() == 1 &&
+  // If the list is not empty at this point, either a decoding error terminated
+  // the while-loop, or list must hold exactly one CNG packet.
+  assert(packet_list->empty() || *decoded_length < 0 ||
+         (packet_list->size() == 1 && packet &&
              decoder_database_->IsComfortNoise(packet->header.payloadType)));
   return 0;
 }
diff --git a/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc
index 1b1e950a82..a8cff5ed3b 100644
--- a/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc
@@ -330,7 +330,7 @@ void NetEqDecodingTest::DecodeAndCompare(const std::string &rtp_file,
     std::ostringstream ss;
     ss << "Lap number " << i++ << " in DecodeAndCompare while loop";
     SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
-    int out_len;
+    int out_len = 0;
     ASSERT_NO_FATAL_FAILURE(Process(&rtp, &out_len));
     ASSERT_NO_FATAL_FAILURE(ref_files.ProcessReference(out_data_, out_len));
   }
@@ -583,7 +583,6 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(TestFrameWaitingTimeStatistics)) {
 
   std::vector<int> waiting_times;
   neteq_->WaitingTimes(&waiting_times);
-  int len = waiting_times.size();
   EXPECT_EQ(num_frames, waiting_times.size());
   // Since all frames are dumped into NetEQ at once, but pulled out with 10 ms
   // spacing (per definition), we expect the delay to increase with 10 ms for
@@ -594,7 +593,7 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(TestFrameWaitingTimeStatistics)) {
 
   // Check statistics again and make sure it's been reset.
   neteq_->WaitingTimes(&waiting_times);
-  len = waiting_times.size();
+  int len = waiting_times.size();
   EXPECT_EQ(0, len);
 
   // Process > 100 frames, and make sure that that we get statistics
diff --git a/webrtc/modules/audio_coding/neteq4/payload_splitter.cc b/webrtc/modules/audio_coding/neteq4/payload_splitter.cc
index e07b61f701..56039a57ec 100644
--- a/webrtc/modules/audio_coding/neteq4/payload_splitter.cc
+++ b/webrtc/modules/audio_coding/neteq4/payload_splitter.cc
@@ -331,7 +331,6 @@ void PayloadSplitter::SplitBySamples(const Packet* packet,
     new_packet->primary = packet->primary;
     new_packet->payload = new uint8_t[len];
     memcpy(new_packet->payload, payload_ptr, len);
-    payload_ptr += len;
     new_packets->push_back(new_packet);
   }
 }
diff --git a/webrtc/modules/audio_coding/neteq4/test/RTPencode.cc b/webrtc/modules/audio_coding/neteq4/test/RTPencode.cc
index 7368b0265d..bc806091b0 100644
--- a/webrtc/modules/audio_coding/neteq4/test/RTPencode.cc
+++ b/webrtc/modules/audio_coding/neteq4/test/RTPencode.cc
@@ -1618,7 +1618,7 @@ int NetEQTest_encode(int coder, int16_t *indata, int frameLen, unsigned char * e
 #ifdef CODEC_G722
         else if (coder==webrtc::kDecoderG722) { /*g722 */
             cdlen=WebRtcG722_Encode(g722EncState[k], indata, frameLen, (int16_t*)encoded);
-            cdlen=frameLen>>1;
+            assert(cdlen == frameLen>>1);
         }
 #endif
 #ifdef CODEC_ILBC
diff --git a/webrtc/modules/audio_coding/neteq4/test/RTPjitter.cc b/webrtc/modules/audio_coding/neteq4/test/RTPjitter.cc
index 79af181fbe..eeb4c90140 100644
--- a/webrtc/modules/audio_coding/neteq4/test/RTPjitter.cc
+++ b/webrtc/modules/audio_coding/neteq4/test/RTPjitter.cc
@@ -64,7 +64,9 @@ int main(int argc, char* argv[])
 	unsigned int	dat_len, rtp_len, Npack, k;
 	arr_time		*time_vec;
 	char			firstline[FIRSTLINELEN];
-	unsigned char	*rtp_vec = NULL, **packet_ptr, *temp_packet;
+	unsigned char* rtp_vec = NULL;
+        unsigned char** packet_ptr = NULL;
+        unsigned char* temp_packet = NULL;
 	const unsigned int kRtpDumpHeaderSize = 4 + 4 + 4 + 2 + 2;
 	uint16_t			len;
 	uint32_t			*offset;
@@ -113,6 +115,11 @@ int main(int argc, char* argv[])
 		dat_len++;
 	}
 	
+  if (dat_len == 0) {
+    fprintf(stderr, "Error: dat_file is empty, no arrival time is given.\n");
+    goto closing;
+  }
+
 	qsort(time_vec,dat_len,sizeof(arr_time),compare_arr_time);
 
 
@@ -146,6 +153,11 @@ int main(int argc, char* argv[])
 		len=(uint16_t) fread(&rtp_vec[rtp_len], sizeof(unsigned char), 2, in_file); // read length of next packet
 	}
 
+	if (Npack == 0) {
+	  fprintf(stderr, "Error: No RTP packet found.\n");
+	  goto closing;
+	}
+
 	packet_ptr = (unsigned char **) malloc(Npack*sizeof(unsigned char*));
 
 	packet_ptr[0]=rtp_vec;
@@ -182,7 +194,10 @@ int main(int argc, char* argv[])
 closing:
 	free(time_vec);
 	free(rtp_vec);
-	fclose(in_file);
+        if (packet_ptr != NULL) {
+	  free(packet_ptr);
+        }
+        fclose(in_file);
 	fclose(dat_file);
 	fclose(out_file);