Fixing Release compilation errors

Review URL: http://webrtc-codereview.appspot.com/267026

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1000 4adac7df-926f-26a2-2b94-8c16560cd09d

src/modules/audio_coding/codecs/g711/test/testG711.cpp

@@ -42,7 +42,7 @@ int readframe(WebRtc_Word16 *data, FILE *inp, int length) {
 int main(int argc, char* argv[])
 {
     char inname[80], outname[40], bitname[40];
-    FILE *inp, *outp, *bitp;
+    FILE *inp, *outp, *bitp = NULL;
     int framecnt, endfile;
 
     WebRtc_Word16 framelength = 80;

src/modules/audio_coding/codecs/g722/test/testG722.cpp

@@ -56,7 +56,7 @@ int main(int argc, char* argv[])
 
     /* Runtime statistics */
     double starttime;
-    double runtime;
+    double runtime = 0;
     double length_file;
 
     WebRtc_Word16 stream_len = 0;

src/modules/audio_coding/codecs/iSAC/fix/test/kenny.c

@@ -101,13 +101,13 @@ int main(int argc, char* argv[])
   WebRtc_Word16 CodingMode;
   WebRtc_Word16 bottleneck;
   WebRtc_Word16 framesize = 30;           /* ms */
-  int cur_framesmpls, err, lostPackets = 0;
+  int cur_framesmpls, err = 0, lostPackets = 0;
 
   /* Runtime statistics */
   double starttime, runtime, length_file;
 
   WebRtc_Word16 stream_len = 0;
-  WebRtc_Word16 framecnt, declen;
+  WebRtc_Word16 framecnt, declen = 0;
   WebRtc_Word16 shortdata[FRAMESAMPLES_10ms];
   WebRtc_Word16 decoded[MAX_FRAMESAMPLES];
   WebRtc_UWord16 streamdata[500];
@@ -119,7 +119,7 @@ int main(int argc, char* argv[])
   WebRtc_Word32 payloadRate = 0;
   int setControlBWE = 0;
   int readLoss;
-  FILE  *plFile;
+  FILE  *plFile = NULL;
 
   char version_number[20];
   char tmpBit[5] = ".bit";
@@ -396,7 +396,9 @@ int main(int argc, char* argv[])
       if (fscanf(f_bn, "%d", &aux_var) == EOF) {
         /* Set pointer to beginning of file */
         fseek(f_bn, 0L, SEEK_SET);
-        fscanf(f_bn, "%d", &aux_var);
+        if (fscanf(f_bn, "%d", &aux_var) == EOF) {
+          exit(0);
+        }
       }
       bottleneck = (WebRtc_Word16)aux_var;
       /* Bottleneck is a cosine function
@@ -640,7 +642,9 @@ int main(int argc, char* argv[])
         if (fscanf(f_bn, "%d", &aux_var) == EOF) {
           /* Set pointer to beginning of file */
           fseek(f_bn, 0L, SEEK_SET);
-          fscanf(f_bn, "%d", &aux_var);
+          if (fscanf(f_bn, "%d", &aux_var) == EOF) {
+            exit(0);
+          }
         }
         bottleneck = (WebRtc_Word16)aux_var;
         if (CodingMode == 1) {
@@ -675,10 +679,11 @@ int main(int argc, char* argv[])
     if (fp_gns != NULL) {
       if (fscanf(fp_gns, "%d", &cur_delay) == EOF) {
         fseek(fp_gns, 0L, SEEK_SET);
-        fscanf(fp_gns, "%d", &cur_delay);
+        if (fscanf(fp_gns, "%d", &cur_delay) == EOF) {
+          exit(0);
+        }
       }
     }
-
 
     /* simulate packet handling through NetEq and the modem */
     if (!(testNum == 3 && framecnt == 0)) {

src/modules/audio_coding/codecs/iSAC/main/test/ReleaseTest-API/ReleaseTest-API.cc

@@ -58,7 +58,7 @@ int main(int argc, char* argv[])
 	double starttime, runtime, length_file;
 
 	WebRtc_Word16 stream_len = 0;
-	WebRtc_Word16 declen, lostFrame = 0, declenTC;
+	WebRtc_Word16 declen, lostFrame = 0, declenTC = 0;
 	
 	WebRtc_Word16 shortdata[SWBFRAMESAMPLES_10ms];
 	WebRtc_Word16 vaddata[SWBFRAMESAMPLES_10ms*3];
@@ -97,9 +97,9 @@ int main(int argc, char* argv[])
     bool doTransCoding = false;
     WebRtc_Word32 rateTransCoding = 0;
     WebRtc_UWord16 streamDataTransCoding[600];
-    WebRtc_Word16 streamLenTransCoding;
+    WebRtc_Word16 streamLenTransCoding = 0;
-    FILE* transCodingFile;
+    FILE* transCodingFile = NULL;
-    FILE* transcodingBitstream;
+    FILE* transcodingBitstream = NULL;
     WebRtc_UWord32 numTransCodingBytes = 0;
 
 	/* only one structure used for ISAC encoder */
@@ -368,7 +368,9 @@ int main(int argc, char* argv[])
                     {
                         /* Set pointer to beginning of file */
                         fseek(f_bn, 0L, SEEK_SET);
-                        fscanf(f_bn, "%d", &bottleneck);
+                        if (fscanf(f_bn, "%d", &bottleneck) == EOF) {
+                            exit(0);
+                        }
                     }		
 
                     /*	Bottleneck is a cosine function 
@@ -744,7 +746,9 @@ int main(int argc, char* argv[])
             {
                 /* Set pointer to beginning of file */
                 fseek(f_bn, 0L, SEEK_SET);
-                fscanf(f_bn, "%d", &bottleneck);
+                if (fscanf(f_bn, "%d", &bottleneck) == EOF) {
+                    exit(0);
+                }
             }
             if(CodingMode == 1)
             {

src/modules/audio_coding/codecs/iSAC/main/test/simpleKenny.c

@@ -108,7 +108,7 @@ int main(int argc, char* argv[])
     WebRtc_UWord16  payload[600];
     WebRtc_UWord16  payloadRCU[600];
     WebRtc_UWord16  packetLossPercent = 0;
-    WebRtc_Word16   rcuStreamLen;
+    WebRtc_Word16   rcuStreamLen = 0;
 	int onlyEncode;
 	int onlyDecode;
 

src/modules/audio_coding/main/source/audio_coding_module.gypi

@@ -100,6 +100,7 @@
           'type': 'executable',
           'dependencies': [
             'audio_coding_module',
+            '<(webrtc_root)/../testing/gtest.gyp:gtest',
             '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
           ],
           'sources': [

src/modules/audio_coding/main/test/APITest.cpp

@@ -9,20 +9,21 @@
  */
 
 #include <cctype>
-#include <stdio.h>
-#include <string.h>
-#include <stdlib.h>
 #include <iostream>
 #include <ostream>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
 
 #include "APITest.h"
-#include "thread_wrapper.h"
+#include "common_types.h"
+#include "engine_configurations.h"
 #include "event_wrapper.h"
+#include "gtest/gtest.h"
+#include "thread_wrapper.h"
 #include "tick_util.h"
 #include "trace.h"
 #include "utility.h"
-#include "common_types.h"
-#include "engine_configurations.h"
 
 #define TEST_DURATION_SEC 600
 
@@ -310,7 +311,7 @@ APITest::SetUp()
     char print[11];
 
     printf("\nRandom Test (y/n)?");
-    fgets(print, 10, stdin); 
+    EXPECT_TRUE(fgets(print, 10, stdin) != NULL);
     print[10] = '\0';
     if(strstr(print, "y") != NULL)
     {
@@ -327,11 +328,11 @@ APITest::SetUp()
         Trace::SetTraceFile("ACMAPITest.txt", true);
         _randomTest = false;
         printf("\nPrint Tests (y/n)? ");
-        fgets(print, 10, stdin); 
+        EXPECT_TRUE(fgets(print, 10, stdin) != NULL);
         print[10] = '\0';
         if(strstr(print, "y") == NULL)
         {
-            freopen("APITest_log.txt", "w", stdout);
+            EXPECT_TRUE(freopen("APITest_log.txt", "w", stdout) != 0);
             _verbose = false;
         }
     }
@@ -1139,8 +1140,8 @@ void
 APITest::TestPlayout(char receiveSide)
 {
     AudioCodingModule* receiveACM;
-    AudioPlayoutMode* playoutMode;
+    AudioPlayoutMode* playoutMode = NULL;
-    ACMBackgroundNoiseMode* bgnMode;
+    ACMBackgroundNoiseMode* bgnMode = NULL;
     switch(receiveSide)
     {
         case 'A':

src/modules/audio_coding/main/test/EncodeDecodeTest.cpp

@@ -9,10 +9,12 @@
  */
 
 #include "EncodeDecodeTest.h"
-#include "common_types.h"
 
 #include <stdlib.h>
 #include <string.h>
+
+#include "common_types.h"
+#include "gtest/gtest.h"
 #include "trace.h"
 #include "utility.h"
 
@@ -63,7 +65,7 @@ void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream)
         printf("\nValid output frequencies:\n");
         printf("8000\n16000\n32000\n-1, which means output freq equal to received signal freq");
         printf("\n\nChoose output sampling frequency: ");
-        scanf("%d", &playSampFreq);
+        ASSERT_GT(scanf("%d", &playSampFreq), 0);
         char fileName[] = "./src/modules/audio_coding/main/test/outFile.pcm";
         _pcmFile.Open(fileName, 32000, "wb+");
     }

src/modules/audio_coding/main/test/EncodeToFileTest.cpp

@@ -9,8 +9,6 @@
  */
 
 #include "EncodeToFileTest.h"
-#include "audio_coding_module.h"
-#include "common_types.h"
 
 #ifdef WIN32
 #   include <Winsock2.h>
@@ -18,11 +16,14 @@
 #   include <arpa/inet.h>
 #endif
 
-
 #include <stdio.h>
 #include <stdlib.h>
 #include <string.h>
 
+#include "audio_coding_module.h"
+#include "common_types.h"
+#include "gtest/gtest.h"
+
 TestPacketization::TestPacketization(RTPStream *rtpStream, WebRtc_UWord16 frequency)
 :
 _frequency(frequency),
@@ -90,8 +91,7 @@ void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream)
             printf("%d %s\n", n, sendCodec.plname);
         }
         printf("Choose your codec:");
-    
+        ASSERT_GT(scanf("%d", &codecNo), 0);
-        scanf("%d", &codecNo);
         char fileName[] = "./test/data/audio_coding/testfile32kHz.pcm";
         _pcmFile.Open(fileName, 32000, "rb");
     }

src/modules/audio_coding/main/test/PCMFile.cpp

@@ -1,3 +1,4 @@
+
 /*
  *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
  *
@@ -8,12 +9,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "PCMFile.h"
+
 #include <cctype>
 #include <stdio.h>
 #include <string.h>
 
-
+#include "gtest/gtest.h"
-#include "PCMFile.h"
 #include "module_common_types.h"
 
 #define MAX_FILE_NAME_LENGTH_BYTE 500
@@ -54,7 +56,7 @@ PCMFile::ChooseFile(
     WebRtc_Word8 tmpName[MAX_FILE_NAME_LENGTH_BYTE];
     //strcpy(_fileName, "in.pcm");
     //printf("\n\nPlease enter the input file: ");
-    fgets(tmpName, MAX_FILE_NAME_LENGTH_BYTE, stdin);
+    EXPECT_TRUE(fgets(tmpName, MAX_FILE_NAME_LENGTH_BYTE, stdin) != NULL);
     tmpName[MAX_FILE_NAME_LENGTH_BYTE-1] = '\0';
     WebRtc_Word16 n = 0;
 
@@ -106,7 +108,7 @@ PCMFile::ChooseFile(
     WebRtc_Word8 tmpName[MAX_FILE_NAME_LENGTH_BYTE];
     //strcpy(_fileName, "in.pcm");
     //printf("\n\nPlease enter the input file: ");
-    fgets(tmpName, MAX_FILE_NAME_LENGTH_BYTE, stdin);
+    EXPECT_TRUE(fgets(tmpName, MAX_FILE_NAME_LENGTH_BYTE, stdin) != NULL);
     tmpName[MAX_FILE_NAME_LENGTH_BYTE-1] = '\0';
     WebRtc_Word16 n = 0;
 
@@ -147,7 +149,7 @@ PCMFile::ChooseFile(
         strncpy(fileName, tmpName, len+1);
     }
     printf("Enter the sampling frequency (in Hz) of the above file [%u]: ", *frequencyHz);
-    fgets(tmpName, 10, stdin);
+    EXPECT_TRUE(fgets(tmpName, 10, stdin) != NULL);
     WebRtc_UWord16 tmpFreq = (WebRtc_UWord16)atoi(tmpName);
     if(tmpFreq > 0)
     {

src/modules/audio_coding/main/test/RTPFile.cpp

@@ -9,8 +9,7 @@
  */
 
 #include "RTPFile.h"
-#include "rw_lock_wrapper.h"
+
-#include "engine_configurations.h"
 #include <stdlib.h>
 
 #ifdef WIN32
@@ -20,6 +19,9 @@
 #endif
 
 #include "audio_coding_module.h"
+#include "engine_configurations.h"
+#include "gtest/gtest.h"
+#include "rw_lock_wrapper.h"
 
 void RTPStream::ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const WebRtc_UWord8* rtpHeader)
 {
@@ -186,16 +188,16 @@ void RTPFile::ReadHeader()
     WebRtc_UWord32 start_sec, start_usec, source;
     WebRtc_UWord16 port, padding;
     char fileHeader[40];
-    fgets(fileHeader, 40, _rtpFile);
+    EXPECT_TRUE(fgets(fileHeader, 40, _rtpFile) != 0);
-    fread(&start_sec, 4, 1, _rtpFile);
+    EXPECT_GT(fread(&start_sec, 4, 1, _rtpFile), 0u);
     start_sec=ntohl(start_sec);
-    fread(&start_usec, 4, 1, _rtpFile);
+    EXPECT_GT(fread(&start_usec, 4, 1, _rtpFile), 0u);
     start_usec=ntohl(start_usec);
-    fread(&source, 4, 1, _rtpFile);
+    EXPECT_GT(fread(&source, 4, 1, _rtpFile), 0u);
     source=ntohl(source);
-    fread(&port, 2, 1, _rtpFile);
+    EXPECT_GT(fread(&port, 2, 1, _rtpFile), 0u);
     port=ntohs(port);
-    fread(&padding, 2, 1, _rtpFile);
+    EXPECT_GT(fread(&padding, 2, 1, _rtpFile), 0u);
     padding=ntohs(padding);
 }
 
@@ -237,19 +239,19 @@ WebRtc_UWord16 RTPFile::Read(WebRtcRTPHeader* rtpInfo,
     WebRtc_UWord16 lengthBytes;
     WebRtc_UWord16 plen;
     WebRtc_UWord8 rtpHeader[12];
-    fread(&lengthBytes, 2, 1, _rtpFile);
+    EXPECT_GT(fread(&lengthBytes, 2, 1, _rtpFile), 0u);
     if (feof(_rtpFile))
     {
         _rtpEOF = true;
         return 0;
     }
-    fread(&plen, 2, 1, _rtpFile);
+    EXPECT_GT(fread(&plen, 2, 1, _rtpFile), 0u);
     if (feof(_rtpFile))
     {
         _rtpEOF = true;
         return 0;
     }
-    fread(offset, 4, 1, _rtpFile);
+    EXPECT_GT(fread(offset, 4, 1, _rtpFile), 0u);
     if (feof(_rtpFile))
     {
         _rtpEOF = true;
@@ -263,7 +265,7 @@ WebRtc_UWord16 RTPFile::Read(WebRtcRTPHeader* rtpInfo,
         throw "Unable to read RTP file";
         exit(1);
     }
-    fread(rtpHeader, 12, 1, _rtpFile);
+    EXPECT_GT(fread(rtpHeader, 12, 1, _rtpFile), 0u);
     if (feof(_rtpFile))
     {
         _rtpEOF = true;
@@ -287,7 +289,7 @@ WebRtc_UWord16 RTPFile::Read(WebRtcRTPHeader* rtpInfo,
         exit(1);
     }
     lengthBytes -= 20;
-    fread(payloadData, 1, lengthBytes, _rtpFile);
+    EXPECT_GT(fread(payloadData, 1, lengthBytes, _rtpFile), 0u);
     if (feof(_rtpFile))
     {
         _rtpEOF = true;

src/modules/audio_coding/main/test/TwoWayCommunication.cpp

@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "TwoWayCommunication.h"
+
 #include <cctype>
 #include <stdio.h>
 #include <string.h>
@@ -16,12 +18,12 @@
 #include <Windows.h>
 #endif
 
-#include "TwoWayCommunication.h"
-#include "engine_configurations.h"
-#include "PCMFile.h"
-#include "utility.h"
-#include "trace.h"
 #include "common_types.h"
+#include "engine_configurations.h"
+#include "gtest/gtest.h"
+#include "PCMFile.h"
+#include "trace.h"
+#include "utility.h"
 
 using namespace webrtc;
 
@@ -79,11 +81,11 @@ TwoWayCommunication::ChooseCodec(WebRtc_UWord8* codecID_A, WebRtc_UWord8* codecI
     }
     printf("\nChoose a send codec for side A [0]: ");
     char myStr[15] = "";
-    fgets(myStr, 10, stdin);
+    EXPECT_TRUE(fgets(myStr, 10, stdin) != NULL);
     *codecID_A = (WebRtc_UWord8)atoi(myStr);
 
     printf("\nChoose a send codec for side B [0]: ");
-    fgets(myStr, 10, stdin);
+    EXPECT_TRUE(fgets(myStr, 10, stdin) != NULL);
     *codecID_B = (WebRtc_UWord8)atoi(myStr);
 
     AudioCodingModule::Destroy(tmpACM);
@@ -97,7 +99,7 @@ TwoWayCommunication::ChooseFile(char* fileName, WebRtc_Word16 maxLen, WebRtc_UWo
     WebRtc_Word8 tmpName[MAX_FILE_NAME_LENGTH_BYTE];
     //strcpy(_fileName, "in.pcm");
     //printf("\n\nPlease enter the input file: ");
-    fgets(tmpName, MAX_FILE_NAME_LENGTH_BYTE, stdin);
+    EXPECT_TRUE(fgets(tmpName, MAX_FILE_NAME_LENGTH_BYTE, stdin) != NULL);
     tmpName[MAX_FILE_NAME_LENGTH_BYTE-1] = '\0';
     WebRtc_Word16 n = 0;
 
@@ -138,7 +140,7 @@ TwoWayCommunication::ChooseFile(char* fileName, WebRtc_Word16 maxLen, WebRtc_UWo
         strncpy(fileName, tmpName, len+1);
     }
     printf("Enter the sampling frequency (in Hz) of the above file [%u]: ", *frequencyHz);
-    fgets(tmpName, 6, stdin);
+    EXPECT_TRUE(fgets(tmpName, 6, stdin) != NULL);
     WebRtc_UWord16 tmpFreq = (WebRtc_UWord16)atoi(tmpName);
     if(tmpFreq > 0)
     {

src/modules/audio_coding/main/test/iSACTest.cpp

@@ -298,7 +298,7 @@ ISACTest::Perform()
     testNr++;
     EncodeDecode(testNr, wbISACConfig, swbISACConfig);
     
-    int dummy;
+    int user_input;
     if((_testMode == 0) || (_testMode == 1))
     {
         swbISACConfig.maxPayloadSizeByte = (WebRtc_UWord16)200;
@@ -307,11 +307,11 @@ ISACTest::Perform()
     else
     {
         printf("Enter the max payload-size for side A: ");
-        scanf("%d", &dummy);
+        CHECK_ERROR(scanf("%d", &user_input));
-        swbISACConfig.maxPayloadSizeByte = (WebRtc_UWord16)dummy;
+        swbISACConfig.maxPayloadSizeByte = (WebRtc_UWord16)user_input;
         printf("Enter the max payload-size for side B: ");
-        scanf("%d", &dummy);
+        CHECK_ERROR(scanf("%d", &user_input));
-        wbISACConfig.maxPayloadSizeByte = (WebRtc_UWord16)dummy;
+        wbISACConfig.maxPayloadSizeByte = (WebRtc_UWord16)user_input;
     }
     testNr++;
     EncodeDecode(testNr, wbISACConfig, swbISACConfig);
@@ -329,11 +329,11 @@ ISACTest::Perform()
     else
     {
         printf("Enter the max rate for side A: ");
-        scanf("%d", &dummy);
+        CHECK_ERROR(scanf("%d", &user_input));
-        swbISACConfig.maxRateBitPerSec = (WebRtc_UWord32)dummy;
+        swbISACConfig.maxRateBitPerSec = (WebRtc_UWord32)user_input;
         printf("Enter the max rate for side B: ");
-        scanf("%d", &dummy);
+        CHECK_ERROR(scanf("%d", &user_input));
-        wbISACConfig.maxRateBitPerSec = (WebRtc_UWord32)dummy;
+        wbISACConfig.maxRateBitPerSec = (WebRtc_UWord32)user_input;
     }
  
     testNr++;

src/modules/audio_coding/main/test/utility.cpp

@@ -9,14 +9,14 @@
  */
 
 #include "utility.h"
+
 #include <assert.h>
 #include <stdio.h>
 #include <stdlib.h>
 
-
 #include "audio_coding_module.h"
 #include "common_types.h"
-
+#include "gtest/gtest.h"
 
 #define NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE 13
 
@@ -128,7 +128,7 @@ ChooseCodec(
     do
     {
         printf("\nChoose a codec [0]: ");
-        fgets(myStr, 10, stdin);
+        EXPECT_TRUE(fgets(myStr, 10, stdin) != NULL);
         codecID = atoi(myStr);
         if((codecID < 0) || (codecID >= noCodec))
         {

src/modules/audio_coding/neteq/neteq.gypi

@@ -177,6 +177,9 @@
         {
           'target_name': 'RTPjitter',
           'type': 'executable',
+          'dependencies': [
+            '<(webrtc_root)/../testing/gtest.gyp:gtest',
+          ],
           'sources': [
             'test/RTPjitter.cc',
           ],
@@ -186,6 +189,7 @@
           'type': 'executable',
           'dependencies': [
             'NetEqTestTools',
+            '<(webrtc_root)/../testing/gtest.gyp:gtest',
           ],
           'sources': [
             'test/RTPanalyze.cc',
@@ -196,6 +200,7 @@
           'type': 'executable',
           'dependencies': [
             'NetEqTestTools',
+            '<(webrtc_root)/../testing/gtest.gyp:gtest',
           ],
           'sources': [
            'test/RTPchange.cc',
@@ -206,6 +211,7 @@
           'type': 'executable',
           'dependencies': [
            'NetEqTestTools',
+            '<(webrtc_root)/../testing/gtest.gyp:gtest',
           ],
           'sources': [
             'test/RTPtimeshift.cc',
@@ -216,6 +222,7 @@
           'type': 'executable',
           'dependencies': [
             'NetEqTestTools',
+            '<(webrtc_root)/../testing/gtest.gyp:gtest',
           ],
           'sources': [
             'test/RTPcat.cc',
@@ -232,6 +239,7 @@
             'iLBC',
             'iSAC',
             'CNG',
+            '<(webrtc_root)/../testing/gtest.gyp:gtest',
           ],
           'direct_dependent_settings': {
             'include_dirs': [

src/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.cc

@@ -21,6 +21,8 @@
 
 #include <cstdlib>
 
+#include "gtest/gtest.h"
+
 #define HDR_SIZE 8 // rtpplay packet header size in bytes
 
 
@@ -145,7 +147,9 @@ int NETEQTEST_RTPpacket::skipFileHeader(FILE *fp)
 
     const int kFirstLineLength = 40;
     char firstline[kFirstLineLength];
-    fgets(firstline, kFirstLineLength, fp);
+    if (fgets(firstline, kFirstLineLength, fp) == NULL) {
+        return -1;
+    }
     if (strncmp(firstline, "#!rtpplay", 9) == 0) {
         if (strncmp(firstline, "#!rtpplay1.0", 12) != 0) {
             return -1;

src/modules/audio_coding/neteq/test/RTPanalyze.cc

@@ -19,8 +19,6 @@
 /* Misc. definitions */
 /*********************/
 
-#define FIRSTLINELEN 40
-
 enum {kRedPayloadType = 127};
 
 int main(int argc, char* argv[])
@@ -44,12 +42,8 @@ int main(int argc, char* argv[])
     // print file header
     fprintf(outFile, "SeqNo  TimeStamp   SendTime  Size    PT  M\n");
 
-
     // read file header 
-	char firstline[FIRSTLINELEN];
+    NETEQTEST_RTPpacket::skipFileHeader(inFile);
-	fgets(firstline, FIRSTLINELEN, inFile);
-	fread(firstline, 4+4+4+2+2, 1, inFile); // start_sec + start_usec	+ source + port + padding
-	
     NETEQTEST_RTPpacket packet;
 
     while (packet.readFromFile(inFile) >= 0)

src/modules/audio_coding/neteq/test/RTPcat.cc

@@ -13,6 +13,7 @@
 #include <vector>
 
 #include "NETEQTEST_RTPpacket.h"
+#include "gtest/gtest.h"
 
 /*********************/
 /* Misc. definitions */
@@ -42,16 +43,18 @@ int main(int argc, char* argv[])
 
     // read file header and write directly to output file
     char firstline[FIRSTLINELEN];
-    fgets(firstline, FIRSTLINELEN, inFile);
+    const unsigned int kRtpDumpHeaderSize = 4 + 4 + 4 + 2 + 2;
-    fputs(firstline, outFile);
+    EXPECT_TRUE(fgets(firstline, FIRSTLINELEN, inFile) != NULL);
-    fread(firstline, 4 + 4 + 4 + 2 + 2, 1, inFile); // start_sec + start_usec	+ source + port + padding
+    EXPECT_GT(fputs(firstline, outFile), 0);
-    fwrite(firstline, 4 + 4 + 4 + 2 + 2, 1, outFile);
+    EXPECT_EQ(kRtpDumpHeaderSize, fread(firstline, 1, kRtpDumpHeaderSize,
+                                        inFile));
+    EXPECT_EQ(kRtpDumpHeaderSize, fwrite(firstline, 1, kRtpDumpHeaderSize,
+                                         outFile));
 
     // close input file and re-open it later (easier to write the loop below)
     fclose(inFile);
 
     for (int i = 1; i < argc - 1; i++) {
-
         inFile = fopen(argv[i], "rb");
         if (!inFile) {
             printf("Cannot open input file %s\n", argv[i]);
@@ -60,28 +63,18 @@ int main(int argc, char* argv[])
         printf("Input RTP file: %s\n", argv[i]);
 
         // skip file header
-        fgets(firstline, FIRSTLINELEN, inFile);
+        NETEQTEST_RTPpacket::skipFileHeader(inFile);
-        fread(firstline, 4 + 4 + 4 + 2 + 2, 1, inFile); // start_sec + start_usec   + source + port + padding
-
         NETEQTEST_RTPpacket packet;
         int packLen = packet.readFromFile(inFile);
         if (packLen < 0) {
             exit(1);
         }
-
         while (packLen >= 0) {
-
             packet.writeToFile(outFile);
-
             packLen = packet.readFromFile(inFile);
-
         }
-
         fclose(inFile);
-
     }
-
     fclose(outFile);
-
     return 0;
 }

src/modules/audio_coding/neteq/test/RTPchange.cc

@@ -13,7 +13,7 @@
 #include <vector>
 
 #include "NETEQTEST_RTPpacket.h"
-
+#include "gtest/gtest.h"
 
 /*********************/
 /* Misc. definitions */
@@ -56,8 +56,11 @@ int main(int argc, char* argv[])
     // read all statistics and insert into map
     // read first line
     char tempStr[100];
-    fgets(tempStr, 100, statFile);
+    if (fgets(tempStr, 100, statFile) == NULL)
-
+    {
+      printf("Failed to read timing file %s\n", argv[2]);
+      return (-1);
+    }
     // define map
     std::map<std::pair<WebRtc_UWord16, WebRtc_UWord32>, WebRtc_UWord32>
         packetStats;
@@ -77,10 +80,24 @@ int main(int argc, char* argv[])
 
     // read file header and write directly to output file
     char firstline[FIRSTLINELEN];
-	fgets(firstline, FIRSTLINELEN, inFile);
+    if (fgets(firstline, FIRSTLINELEN, inFile) == NULL)
+    {
+      printf("Failed to read first line of input file %s\n", argv[1]);
+      return (-1);
+    }
     fputs(firstline, outFile);
-	fread(firstline, 4+4+4+2+2, 1, inFile); // start_sec + start_usec	+ source + port + padding
+    // start_sec + start_usec + source + port + padding
-	fwrite(firstline, 4+4+4+2+2, 1, outFile);
+    const unsigned int kRtpDumpHeaderSize = 4 + 4 + 4 + 2 + 2;
+    if (fread(firstline, 1, kRtpDumpHeaderSize, inFile) != kRtpDumpHeaderSize)
+    {
+      printf("Failed to read RTP dump header from input file %s\n", argv[1]);
+      return (-1);
+    }
+    if (fwrite(firstline, 1, kRtpDumpHeaderSize, outFile) != kRtpDumpHeaderSize)
+    {
+      printf("Failed to write RTP dump header to output file %s\n", argv[3]);
+      return (-1);
+    }
 
     std::vector<NETEQTEST_RTPpacket *> packetVec;
 

src/modules/audio_coding/neteq/test/RTPencode.cc

@@ -277,7 +277,7 @@ int main(int argc, char* argv[])
     bool dtmfSent = false;
 #endif
     bool usingStereo = false;
-    int stereoMode;
+    int stereoMode = 0;
     int numChannels = 1;
 
 	/* check number of parameters */
@@ -1609,7 +1609,7 @@ int NetEQTest_free_coders(enum WebRtcNetEQDecoder coder, int numChannels) {
 int NetEQTest_encode(int coder, WebRtc_Word16 *indata, int frameLen, unsigned char * encoded,int sampleRate , 
 						  int * vad, int useVAD, int bitrate, int numChannels){
 
-	short cdlen;
+	short cdlen = 0;
 	WebRtc_Word16 *tempdata;
 	static int first_cng=1;
 	WebRtc_Word16 tempLen;

src/modules/audio_coding/neteq/test/RTPjitter.cc

@@ -22,6 +22,8 @@
 #include <search.h>
 #include <float.h>
 
+#include "gtest/gtest.h"
+
 /*********************/
 /* Misc. definitions */
 /*********************/
@@ -68,7 +70,8 @@ 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, **packet_ptr, *temp_packet;
+	unsigned char	*rtp_vec = NULL, **packet_ptr, *temp_packet;
+	const unsigned int kRtpDumpHeaderSize = 4 + 4 + 4 + 2 + 2;
 	WebRtc_UWord16			len;
 	WebRtc_UWord32			*offset;
 
@@ -126,10 +129,12 @@ int main(int argc, char* argv[])
 	}
 
 	// read file header and write directly to output file
-	fgets(firstline, FIRSTLINELEN, in_file);
+	EXPECT_TRUE(fgets(firstline, FIRSTLINELEN, in_file) != NULL);
-	fputs(firstline, out_file);
+	EXPECT_GT(fputs(firstline, out_file), 0);
-	fread(firstline, 4+4+4+2+2, 1, in_file); // start_sec + start_usec	+ source + port + padding
+	EXPECT_EQ(kRtpDumpHeaderSize, fread(firstline, 1, kRtpDumpHeaderSize,
-	fwrite(firstline, 4+4+4+2+2, 1, out_file);
+	                                    in_file));
+	EXPECT_EQ(kRtpDumpHeaderSize, fwrite(firstline, 1, kRtpDumpHeaderSize,
+	                                     out_file));
 
 	// read all RTP packets into vector
 	rtp_len=0;

src/modules/audio_coding/neteq/test/RTPtimeshift.cc

@@ -13,7 +13,7 @@
 #include <vector>
 
 #include "NETEQTEST_RTPpacket.h"
-
+#include "gtest/gtest.h"
 
 /*********************/
 /* Misc. definitions */
@@ -47,12 +47,14 @@ int main(int argc, char* argv[])
 	printf("Output RTP file: %s\n\n",argv[2]);
 
     // read file header and write directly to output file
+	const unsigned int kRtpDumpHeaderSize = 4 + 4 + 4 + 2 + 2;
 	char firstline[FIRSTLINELEN];
-	fgets(firstline, FIRSTLINELEN, inFile);
+	EXPECT_TRUE(fgets(firstline, FIRSTLINELEN, inFile) != NULL);
-	fputs(firstline, outFile);
+	EXPECT_GT(fputs(firstline, outFile), 0);
-	fread(firstline, 4+4+4+2+2, 1, inFile); // start_sec + start_usec	+ source + port + padding
+	EXPECT_EQ(kRtpDumpHeaderSize,
-	fwrite(firstline, 4+4+4+2+2, 1, outFile);
+	          fread(firstline, 1, kRtpDumpHeaderSize, inFile));
-
+	EXPECT_EQ(kRtpDumpHeaderSize,
+	          fwrite(firstline, 1, kRtpDumpHeaderSize, outFile));
 	NETEQTEST_RTPpacket packet;
 	int packLen = packet.readFromFile(inFile);
 	if (packLen < 0)

src/modules/audio_device/main/test/func_test_manager.cc

@@ -776,7 +776,7 @@ WebRtc_Word32 FuncTestManager::TestAudioLayerSelection()
     {
         TEST_LOG("Would you like to try kWindowsCoreAudio instead "
             "[requires Win Vista or Win 7] (Y/N)?\n: ");
-        scanf(" %c", &ch);
+        TEST(scanf(" %c", &ch) > 0);
         ch = toupper(ch);
         if (ch == 'Y')
         {
@@ -785,7 +785,7 @@ WebRtc_Word32 FuncTestManager::TestAudioLayerSelection()
     } else if (audioLayer == AudioDeviceModule::kWindowsCoreAudio)
     {
         TEST_LOG("Would you like to try kWindowsWaveAudio instead (Y/N)?\n: ");
-        scanf(" %c", &ch);
+        TEST(scanf(" %c", &ch) > 0);
         ch = toupper(ch);
         if (ch == 'Y')
         {
@@ -1695,7 +1695,7 @@ WebRtc_Word32 FuncTestManager::TestMicrophoneVolume()
              RecordedMicrophoneVolumeFile);
     char ch;
     bool fileRecording(false);
-    scanf(" %c", &ch);
+    TEST(scanf(" %c", &ch) > 0);
     ch = toupper(ch);
     if (ch == 'Y')
     {
@@ -1834,7 +1834,7 @@ WebRtc_Word32 FuncTestManager::TestMicrophoneMute()
         RecordedMicrophoneMuteFile);
     char ch;
     bool fileRecording(false);
-    scanf(" %c", &ch);
+    TEST(scanf(" %c", &ch) > 0);
     ch = toupper(ch);
     if (ch == 'Y')
     {
@@ -1970,7 +1970,7 @@ WebRtc_Word32 FuncTestManager::TestMicrophoneBoost()
         RecordedMicrophoneBoostFile);
     char ch;
     bool fileRecording(false);
-    scanf(" %c", &ch);
+    TEST(scanf(" %c", &ch) > 0);
     ch = toupper(ch);
     if (ch == 'Y')
     {
@@ -2107,7 +2107,7 @@ WebRtc_Word32 FuncTestManager::TestMicrophoneAGC()
         RecordedMicrophoneAGCFile);
     char ch;
     bool fileRecording(false);
-    scanf(" %c", &ch);
+    TEST(scanf(" %c", &ch) > 0);
     ch = toupper(ch);
     if (ch == 'Y')
     {
@@ -2567,11 +2567,8 @@ WebRtc_Word32 FuncTestManager::SelectRecordingDevice()
         TEST_LOG(" (%d) Device %d (%s)\n", i, i, name);
     }
     TEST_LOG("\n: ");
-
     int sel(0);
-
+    TEST(scanf("%u", &sel) > 0);
-    scanf("%u", &sel);
-
     if (sel < (nDevices))
     {
         TEST((ret = _audioDevice->SetRecordingDevice(sel)) == 0);
@@ -2634,13 +2631,9 @@ WebRtc_Word32 FuncTestManager::SelectPlayoutDevice()
         TEST_LOG(" (%d) Device %d (%s)\n", i, i, name);
     }
     TEST_LOG("\n: ");
-
     int sel(0);
-
+    TEST(scanf("%u", &sel) > 0);
-    scanf("%u", &sel);
-
     WebRtc_Word32 ret(0);
-
     if (sel < (nDevices))
     {
         TEST((ret = _audioDevice->SetPlayoutDevice(sel)) == 0);

src/modules/video_coding/codecs/test_framework/normal_async_test.cc

@@ -9,12 +9,15 @@
  */
 
 #include "normal_async_test.h"
-#include "typedefs.h"
+
 #include <sstream>
 #include <assert.h>
 #include <queue>
 #include <string.h>
+
+#include "gtest/gtest.h"
 #include "tick_util.h"
+#include "typedefs.h"
 
 using namespace webrtc;
 
@@ -391,7 +394,7 @@ bool
 NormalAsyncTest::Encode()
 {
     _lengthEncFrame = 0;
-    fread(_sourceBuffer, 1, _lengthSourceFrame, _sourceFile);
+    EXPECT_GT(fread(_sourceBuffer, 1, _lengthSourceFrame, _sourceFile), 0u);
     _inputVideoBuffer.CopyBuffer(_lengthSourceFrame, _sourceBuffer);
     _inputVideoBuffer.SetTimeStamp((unsigned int)
         (_encFrameCnt * 9e4 / _inst.maxFramerate));
@@ -448,6 +451,7 @@ NormalAsyncTest::Encode()
 
     webrtc::CodecSpecificInfo* codecSpecificInfo = CreateEncoderSpecificInfo();
     int ret = _encoder->Encode(rawImage, codecSpecificInfo, &frameType);
+    EXPECT_EQ(ret, WEBRTC_VIDEO_CODEC_OK);
     if (codecSpecificInfo != NULL)
     {
         delete codecSpecificInfo;

src/modules/video_coding/codecs/test_framework/normal_test.cc

@@ -9,10 +9,13 @@
  */
 
 #include "normal_test.h"
+
 #include <time.h>
 #include <sstream>
 #include <string.h>
 
+#include "gtest/gtest.h"
+
 NormalTest::NormalTest()
 :
 Test("Normal Test 1", "A test of normal execution of the codec"),
@@ -169,7 +172,7 @@ bool
 NormalTest::Encode()
 {
     _lengthEncFrame = 0;
-    fread(_sourceBuffer, 1, _lengthSourceFrame, _sourceFile);
+    EXPECT_GT(fread(_sourceBuffer, 1, _lengthSourceFrame, _sourceFile), 0u);
     if (feof(_sourceFile) != 0)
     {
         return true;

src/modules/video_coding/codecs/test_framework/performance_test.cc

@@ -9,9 +9,12 @@
  */
 
 #include "performance_test.h"
-#include "tick_util.h"
+
 #include <assert.h>
 
+#include "gtest/gtest.h"
+#include "tick_util.h"
+
 using namespace webrtc;
 
 #define NUM_FRAMES 300
@@ -129,7 +132,8 @@ PerformanceTest::Perform()
             // Read a new frame from file
             WriteLockScoped imageLock(*_rawImageLock);
             _lengthEncFrame = 0;
-            fread(_sourceBuffer, 1, _lengthSourceFrame, _sourceFile);
+            EXPECT_GT(fread(_sourceBuffer, 1, _lengthSourceFrame, _sourceFile),
+                      0u);
             if (feof(_sourceFile) != 0)
             {
                 rewind(_sourceFile);
@@ -269,6 +273,7 @@ bool PerformanceTest::Encode()
     }
     webrtc::CodecSpecificInfo* codecSpecificInfo = CreateEncoderSpecificInfo();
     int ret = _encoder->Encode(rawImage, codecSpecificInfo, &frameType);
+    EXPECT_EQ(ret, WEBRTC_VIDEO_CODEC_OK);
     if (codecSpecificInfo != NULL)
     {
         delete codecSpecificInfo;

src/modules/video_coding/codecs/test_framework/test_framework.gypi

@@ -16,6 +16,7 @@
           'type': '<(library)',
 
           'dependencies': [
+            '<(webrtc_root)/../testing/gtest.gyp:gtest',
             '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
             '<(webrtc_root)/common_video/common_video.gyp:webrtc_vplib',
           ],

src/modules/video_coding/main/source/video_coding_test.gypi

@@ -11,6 +11,7 @@
       'target_name': 'video_coding_test',
       'type': 'executable',
       'dependencies': [
+         '<(webrtc_root)/../testing/gtest.gyp:gtest',
          '<(webrtc_root)/../test/test.gyp:test_support',
          'webrtc_video_coding',
          'rtp_rtcp',

src/modules/video_coding/main/test/codec_database_test.cc

@@ -121,7 +121,7 @@ CodecDataBaseTest::Perform(CmdArgs& args)
     VideoFrame sourceFrame;
     sourceFrame.VerifyAndAllocate(_lengthSourceFrame);
     WebRtc_UWord8* tmpBuffer = new WebRtc_UWord8[_lengthSourceFrame];
-    fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile);
+    TEST(fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) > 0);
     sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);
     sourceFrame.SetHeight(_height);
     sourceFrame.SetWidth(_width);
@@ -324,7 +324,7 @@ CodecDataBaseTest::Perform(CmdArgs& args)
             for (j=0; j < int(300/VideoCodingModule::NumberOfCodecs()); j++)// assuming 300 frames, NumberOfCodecs <= 10
             {
                 frameCnt++;
-                fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile);
+                TEST(fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) > 0);
                 // building source frame
                 sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);
                 sourceFrame.SetHeight(_height);

src/modules/video_coding/main/test/generic_codec_test.cc

@@ -143,7 +143,7 @@ GenericCodecTest::Perform(CmdArgs& args)
         _vcm->RegisterReceiveCodec(&receiveCodec, 1);
     }
     WebRtc_UWord8* tmpBuffer = new WebRtc_UWord8[_lengthSourceFrame];
-    fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile);
+    TEST(fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) > 0);
     // building source frame
     sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);
     sourceFrame.SetHeight(_height);
@@ -191,7 +191,7 @@ GenericCodecTest::Perform(CmdArgs& args)
     //encoding 1 second of video
     for (i = 0; i < _frameRate; i++)
     {
-        fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile);
+        TEST(fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) > 0);
         sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);
         sourceFrame.SetHeight(_height);
         sourceFrame.SetWidth(_width);
@@ -265,7 +265,7 @@ GenericCodecTest::Perform(CmdArgs& args)
     sourceFrame.VerifyAndAllocate(_lengthSourceFrame);
     const float bitRate[] = {100, 400, 600, 1000, 2000, 3000};
     const float nBitrates = sizeof(bitRate)/sizeof(*bitRate);
-    float _bitRate;
+    float _bitRate = 0;
     int _frameCnt = 0;
     WebRtc_Word64 startTime, currentTime, oneSecTime;
     float totalBytesOneSec;//, totalBytesTenSec;
@@ -377,7 +377,7 @@ GenericCodecTest::Perform(CmdArgs& args)
         encodeComplete = false;
         while (encodeComplete == false)
         {
-            fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile);
+            TEST(fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) > 0);
             _frameCnt++;
             sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);
             sourceFrame.SetHeight(_height);
@@ -467,7 +467,7 @@ GenericCodecTest::Perform(CmdArgs& args)
     rewind(_sourceFile);
     while (!feof(_sourceFile))
     {
-        fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile);
+        TEST(fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) > 0);
         sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);
         sourceFrame.SetHeight(_height);
         sourceFrame.SetWidth(_width);

src/modules/video_coding/main/test/jitter_buffer_test.cc

@@ -102,7 +102,7 @@ int JitterBufferTest(CmdArgs& args)
         packet.seqNum = seqNum;
         packet.payloadType = 126;
         seqNum++;
-        fb->InsertPacket(packet, VCMTickTime::MillisecondTimestamp());
+        fb->InsertPacket(packet, VCMTickTime::MillisecondTimestamp(), false, 0);
         TEST(frameList.Insert(fb) == 0);
     }
     VCMFrameListItem* item = NULL;
@@ -1907,8 +1907,8 @@ int JitterBufferTest(CmdArgs& args)
     seqNum = 65485;
     timeStampStart = timeStamp +  33*90;
     WebRtc_UWord32 timeStampFirstKey = 0;
-    VCMEncodedFrame* ptrLastDeltaFrame;
+    VCMEncodedFrame* ptrLastDeltaFrame = NULL;
-    VCMEncodedFrame* ptrFirstKeyFrame;
+    VCMEncodedFrame* ptrFirstKeyFrame = NULL;
     // insert MAX_NUMBER_OF_FRAMES frames
     do
     {

src/modules/video_coding/main/test/media_opt_test.cc

@@ -128,7 +128,7 @@ MediaOptTest::Setup(int testType, CmdArgs& args)
         _fpinp = fopen("dat_inp","rb");
         _fpout = fopen("test_runs/dat_out","ab");
         _fpout2 = fopen("test_runs/dat_out2","ab");
-        fscanf(_fpinp,"%f %f %d \n",&rateTest,&lossTest,&numRuns);
+        TEST(fscanf(_fpinp,"%f %f %d \n",&rateTest,&lossTest,&numRuns) > 0);
         _bitRate = rateTest;
         _lossRate = lossTest;
         _testNum = 0;
@@ -311,7 +311,7 @@ MediaOptTest::Perform()
 
     while (feof(_sourceFile)== 0)
     {
-        fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile);
+        TEST(fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) > 0);
         _frameCnt++;
 
         sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);

src/modules/video_coding/main/test/mt_rx_tx_test.cc

@@ -55,7 +55,7 @@ MainSenderThread(void* obj)
     }
     if (feof(state->_sourceFile) == 0)
     {
-        fread(tmpBuffer, 1, lengthSourceFrame,state->_sourceFile);
+        TEST(fread(tmpBuffer, 1, lengthSourceFrame,state->_sourceFile) > 0);
         state->_frameCnt++;
         sourceFrame.CopyFrame(lengthSourceFrame, tmpBuffer);
         sourceFrame.SetHeight(height);

src/modules/video_coding/main/test/normal_test.cc

@@ -279,7 +279,7 @@ NormalTest::Perform(CmdArgs& args)
 #if !(defined(TICK_TIME_DEBUG) || defined(EVENT_DEBUG))
         WebRtc_Word64 processStartTime = VCMTickTime::MillisecondTimestamp();
 #endif
-        fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile);
+        TEST(fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) > 0);
         _frameCnt++;
         sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);
         sourceFrame.SetHeight(_height);

src/modules/video_coding/main/test/quality_modes_test.cc

@@ -236,7 +236,7 @@ QualityModesTest::Perform()
 
     while (feof(_sourceFile)== 0)
     {
-        fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile);
+        TEST(fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) > 0);
         _frameCnt++;
         sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);
         sourceFrame.SetHeight(_nativeHeight);

src/modules/video_coding/main/test/rtp_player.cc

@@ -9,9 +9,6 @@
  */
 
 #include "rtp_player.h"
-#include "../source/internal_defines.h"
-#include "rtp_rtcp.h"
-#include "tick_time.h"
 
 #include <cstdlib>
 #ifdef WIN32
@@ -21,6 +18,11 @@
 #include <arpa/inet.h>
 #endif
 
+#include "../source/internal_defines.h"
+#include "gtest/gtest.h"
+#include "rtp_rtcp.h"
+#include "tick_time.h"
+
 using namespace webrtc;
 
 RawRtpPacket::RawRtpPacket(WebRtc_UWord8* data, WebRtc_UWord16 len)
@@ -231,7 +233,7 @@ WebRtc_Word32 RTPPlayer::ReadHeader()
     {
         return -1;
     }
-    fgets(firstline, FIRSTLINELEN, _rtpFile);
+    EXPECT_TRUE(fgets(firstline, FIRSTLINELEN, _rtpFile) != NULL);
     if(strncmp(firstline,"#!rtpplay",9) == 0) {
         if(strncmp(firstline,"#!rtpplay1.0",12) != 0){
             printf("ERROR: wrong rtpplay version, must be 1.0\n");
@@ -255,15 +257,15 @@ WebRtc_Word32 RTPPlayer::ReadHeader()
     WebRtc_UWord16 port;
     WebRtc_UWord16 padding;
 
-    fread(&start_sec, 4, 1, _rtpFile);
+    EXPECT_GT(fread(&start_sec, 4, 1, _rtpFile), 0u);
     start_sec=ntohl(start_sec);
-    fread(&start_usec, 4, 1, _rtpFile);
+    EXPECT_GT(fread(&start_usec, 4, 1, _rtpFile), 0u);
     start_usec=ntohl(start_usec);
-    fread(&source, 4, 1, _rtpFile);
+    EXPECT_GT(fread(&source, 4, 1, _rtpFile), 0u);
     source=ntohl(source);
-    fread(&port, 2, 1, _rtpFile);
+    EXPECT_GT(fread(&port, 2, 1, _rtpFile), 0u);
     port=ntohs(port);
-    fread(&padding, 2, 1, _rtpFile);
+    EXPECT_GT(fread(&padding, 2, 1, _rtpFile), 0u);
     padding=ntohs(padding);
     return 0;
 }

src/modules/video_processing/main/test/unit_test/unit_test.cc

@@ -298,7 +298,7 @@ TEST_F(VideoProcessingModuleTest, Resampler)
   // Reading test frame
   VideoFrame sourceFrame;
   ASSERT_EQ(0, sourceFrame.VerifyAndAllocate(lengthSourceFrame));
-  fread(sourceFrame.Buffer(), 1, lengthSourceFrame, _sourceFile);
+  EXPECT_GT(fread(sourceFrame.Buffer(), 1, lengthSourceFrame, _sourceFile), 0u);
   ASSERT_EQ(0, sourceFrame.SetLength(lengthSourceFrame));
   sourceFrame.SetHeight(height);
   sourceFrame.SetWidth(width);

webrtc.gyp

@@ -17,8 +17,7 @@
       'dependencies': [
         'src/common_audio/common_audio.gyp:*',
         'src/common_video/common_video.gyp:*',
-        # TODO(andrew): enable when Linux-Release errors are fixed.
+        'src/modules/modules.gyp:*',
-        #'src/modules/modules.gyp:*',
         'src/system_wrappers/source/system_wrappers.gyp:*',
         'src/video_engine/video_engine.gyp:*',
         'src/voice_engine/voice_engine.gyp:*',