diff --git a/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc b/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
index ca05c2ce2..db346026b 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
@@ -44,7 +44,7 @@ int main(int argc, char* argv[])
 {
 
     char inname[100], outname[100], bottleneck_file[100], vadfile[100];
-	FILE *inp, *outp, *f_bn=NULL, *vadp, *bandwidthp;
+	FILE *inp, *outp, *f_bn = NULL, *vadp = NULL, *bandwidthp;
 	int framecnt, endfile;
 
 	int i, errtype, VADusage = 0, packetLossPercent = 0;
@@ -100,8 +100,8 @@ int main(int argc, char* argv[])
     WebRtc_UWord32 numTransCodingBytes = 0;
 
 	/* only one structure used for ISAC encoder */
-	ISACStruct* ISAC_main_inst;
-    ISACStruct* decoderTransCoding;
+	ISACStruct* ISAC_main_inst = NULL;
+    ISACStruct* decoderTransCoding = NULL;
 
 	BottleNeckModel       BN_data;
 
diff --git a/webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc b/webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc
index d6cd8beaa..1918ff800 100644
--- a/webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc
+++ b/webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc
@@ -68,7 +68,7 @@ int main() {
   ForwardErrorCorrection::RecoveredPacketList recoveredPacketList;
   std::list<WebRtc_UWord8*> fecMaskList;
 
-  ForwardErrorCorrection::Packet* mediaPacket;
+  ForwardErrorCorrection::Packet* mediaPacket = NULL;
   // Running over only one loss rate to limit execution time.
   const float lossRate[] = {0.5f};
   const WebRtc_UWord32 lossRateSize = sizeof(lossRate)/sizeof(*lossRate);