Fixed GCC 4.6 errors (mostly 'unused variable' errors and incorrect usage of EXPECT_EQ with booleans.

Fixed remaining compilation errors in release, etc.

Fixed errors from GCC 4.6 compilation.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/366008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1554 4adac7df-926f-26a2-2b94-8c16560cd09d

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

@@ -52,7 +52,6 @@ int main(int argc, char* argv[])
 	WebRtc_Word32 bottleneck;
 	WebRtc_Word16 framesize = 30;           /* ms */
 	int cur_framesmpls, err;
-	int testCE=0;
 
 	/* Runtime statistics */
 	double starttime, runtime, length_file;
@@ -73,7 +72,6 @@ int main(int argc, char* argv[])
     int setControlBWE = 0;
     short FL, testNum;
 	char version_number[20];
-	int readLoss; 
     FILE  *plFile;
     WebRtc_Word32 sendBN;
 
@@ -86,7 +84,6 @@ int main(int argc, char* argv[])
 
     /* If use GNS file */
     FILE *fp_gns = NULL;
-	int gns = 0;
 	char gns_file[100];
     short maxStreamLen30 = 0;
     short maxStreamLen60 = 0;
@@ -180,7 +177,6 @@ int main(int argc, char* argv[])
     /* Loop over all command line arguments */
 	CodingMode = 0;
 	testNum = 0;
-	testCE = 0;
     useAssign = 0;
     //logFile = NULL;
     char transCodingFileName[500];
@@ -308,11 +304,9 @@ int main(int argc, char* argv[])
                 {
 					printf( "\nNo Packet Loss Is Simulated \n" );
                 }
-				readLoss = 0;
             }
             else
             {
-				readLoss = 1;
 				plFile = fopen( argv[i+1], "rb" );
 				if( plFile == NULL )
                 {
@@ -341,7 +335,6 @@ int main(int argc, char* argv[])
 				printf("Cannot read file %s.\n", gns_file);
 				exit(0);
 			}
-			gns = 1;
 			i++;
 		}
 
@@ -642,8 +635,6 @@ int main(int argc, char* argv[])
 		cur_framesmpls = 0;
 		while (1)
         {
-            int kkk;
-            
             /* Read 10 ms speech block */
             endfile = readframe(shortdata, inp, samplesIn10Ms);
 
@@ -681,7 +672,7 @@ int main(int argc, char* argv[])
                     cout << flush;
                 }
 
-                kkk = WebRtcIsac_GetUplinkBw(ISAC_main_inst, &sendBN);
+                WebRtcIsac_GetUplinkBw(ISAC_main_inst, &sendBN);
 
                 if(stream_len>0)
                 {
@@ -723,7 +714,6 @@ int main(int argc, char* argv[])
             }
             else
             {
-                kkk = 0;
                 break;
             }
 

src/modules/audio_coding/codecs/iSAC/main/test/SwitchingSampRate/SwitchingSampRate.cc

@@ -58,7 +58,6 @@ int main(int argc, char* argv[])
   BottleNeckModel* packetData[MAX_NUM_CLIENTS];
 
   char versionNumber[100];
-  short bottleneckChange[MAX_NUM_CLIENTS];
   short samplesIn10ms[MAX_NUM_CLIENTS];
   int bottleneck[MAX_NUM_CLIENTS];
 
@@ -108,7 +107,6 @@ int main(int argc, char* argv[])
            (encoderSampRate[clientCntr] == kIsacWideband)
            ? "wideband":"super-wideband");
 
-    bottleneckChange[clientCntr] = 2 * clientCntr - 1;
     // Open output File Name
     OPEN_FILE_WB(outFile[clientCntr], outFileName[clientCntr]);
     printf("Output File...................... %s\n", outFileName[clientCntr]);

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

@@ -291,7 +291,6 @@ void EncodeDecodeTest::Perform() {
 
   int numCodecs = 1;
   int codePars[3]; //freq, pacsize, rate
-  int playoutFreq[3]; //8, 16, 32k
   int numPars[52]; //number of codec parameters sets (rate,freq,pacsize)to test,
                    //for a given codec
 
@@ -318,7 +317,6 @@ void EncodeDecodeTest::Perform() {
       }
     }
     AudioCodingModule::Destroy(acmTmp);
-    playoutFreq[1] = 16000;
   } else if (_testMode == 0) {
     AudioCodingModule *acmTmp = AudioCodingModule::Create(0);
     numCodecs = acmTmp->NumberOfCodecs();
@@ -337,11 +335,9 @@ void EncodeDecodeTest::Perform() {
         numPars[i] = 0;
       }
     }
-    playoutFreq[1] = 16000;
   } else {
     numCodecs = 1;
     numPars[0] = 1;
-    playoutFreq[1]=16000;
   }
 
   _receiver.testMode = _testMode;

src/modules/audio_coding/main/test/TestVADDTX.cc

@@ -53,7 +53,6 @@ TestVADDTX::~TestVADDTX()
 
 void TestVADDTX::Perform()
 {
-
     if(_testMode == 0)
     {
         printf("Running VAD/DTX Test");
@@ -213,12 +212,15 @@ void TestVADDTX::runTestInternalDTX()
 
 void TestVADDTX::SetVAD(bool statusDTX, bool statusVAD, WebRtc_Word16 vadMode)
 {
-    WebRtc_Word32 status;
     bool dtxEnabled, vadEnabled;
     ACMVADMode vadModeSet;
 
-    status = _acmA->SetVAD(statusDTX, statusVAD, (ACMVADMode) vadMode);
+    if (_acmA->SetVAD(statusDTX, statusVAD, (ACMVADMode) vadMode) < 0) {
-    status = _acmA->VAD(dtxEnabled, vadEnabled, vadModeSet);
+      assert(false);
+    }
+    if (_acmA->VAD(dtxEnabled, vadEnabled, vadModeSet) < 0) {
+      assert(false);
+    }
 
     if(_testMode != 0)
     {
@@ -256,11 +258,12 @@ void TestVADDTX::SetVAD(bool statusDTX, bool statusVAD, WebRtc_Word16 vadMode)
 VADDTXstruct TestVADDTX::GetVAD()
 {
     VADDTXstruct retStruct;
-    WebRtc_Word32 status;
     bool dtxEnabled, vadEnabled;
     ACMVADMode vadModeSet;
 
-    status = _acmA->VAD(dtxEnabled, vadEnabled, vadModeSet);
+    if (_acmA->VAD(dtxEnabled, vadEnabled, vadModeSet) < 0) {
+      assert(false);
+    }
 
     retStruct.statusDTX = dtxEnabled;
     retStruct.statusVAD = vadEnabled;

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

@@ -77,7 +77,6 @@ int func_test(int sel)
         TEST_LOG("- - - - - - - - - - - - - - - - - - - -\n");
         TEST_LOG("\n: ");
 
-        int dummy(0);
         int selection(0);
         enum TestType testType(TTInvalid);
 
@@ -89,7 +88,9 @@ SHOW_MENU:
         }
         else
         {
-            dummy = scanf("%d", &selection);
+            if (scanf("%d", &selection) < 0) {
+              perror("Failed to get selection.");
+            }
         }
 
         switch (selection)

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

@@ -664,58 +664,56 @@ WebRtc_Word32 FuncTestManager::Close()
 
 WebRtc_Word32 FuncTestManager::DoTest(const TestType testType)
 {
-    WebRtc_UWord32 ret(0);
-
     switch (testType)
     {
         case TTAll:
-            ret = TestAudioLayerSelection();
+            TestAudioLayerSelection();
-            ret = TestDeviceEnumeration();
+            TestDeviceEnumeration();
-            ret = TestDeviceSelection();
+            TestDeviceSelection();
-            ret = TestAudioTransport();
+            TestAudioTransport();
-            ret = TestSpeakerVolume();
+            TestSpeakerVolume();
-            ret = TestMicrophoneVolume();
+            TestMicrophoneVolume();
-            ret = TestLoopback();
+            TestLoopback();
         case TTAudioLayerSelection:
             TestAudioLayerSelection();
             break;
         case TTDeviceEnumeration:
-            ret = TestDeviceEnumeration();
+            TestDeviceEnumeration();
             break;
         case TTDeviceSelection:
-            ret = TestDeviceSelection();
+            TestDeviceSelection();
             break;
         case TTAudioTransport:
-            ret = TestAudioTransport();
+            TestAudioTransport();
             break;
         case TTSpeakerVolume:
-            ret = TestSpeakerVolume();
+            TestSpeakerVolume();
             break;
         case TTMicrophoneVolume:
-            ret = TestMicrophoneVolume();
+            TestMicrophoneVolume();
             break;
         case TTSpeakerMute:
-            ret = TestSpeakerMute();
+            TestSpeakerMute();
             break;
         case TTMicrophoneMute:
-            ret = TestMicrophoneMute();
+            TestMicrophoneMute();
             break;
         case TTMicrophoneBoost:
-            ret = TestMicrophoneBoost();
+            TestMicrophoneBoost();
             break;
         case TTMicrophoneAGC:
-            ret = TestMicrophoneAGC();
+            TestMicrophoneAGC();
             break;
         case TTLoopback:
-            ret = TestLoopback();
+            TestLoopback();
             break;
         case TTDeviceRemoval:
-            ret = TestDeviceRemoval();
+            TestDeviceRemoval();
             break;
         case TTMobileAPI:
-            ret = TestAdvancedMBAPI();
+            TestAdvancedMBAPI();
         case TTTest:
-            ret = TestExtra();
+            TestExtra();
             break;
         default:
             break;

src/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc

@@ -88,11 +88,11 @@ void RtcpFormatRembTest::TearDown() {
 }
 
 TEST_F(RtcpFormatRembTest, TestBasicAPI) {
-  EXPECT_EQ(false, rtcp_sender_->REMB());
+  EXPECT_FALSE(rtcp_sender_->REMB());
   EXPECT_EQ(0, rtcp_sender_->SetREMBStatus(true));
-  EXPECT_EQ(true, rtcp_sender_->REMB());
+  EXPECT_TRUE(rtcp_sender_->REMB());
   EXPECT_EQ(0, rtcp_sender_->SetREMBStatus(false));
-  EXPECT_EQ(false, rtcp_sender_->REMB());
+  EXPECT_FALSE(rtcp_sender_->REMB());
 
   EXPECT_EQ(0, rtcp_sender_->SetREMBData(1234, 0, NULL));
 }

src/modules/rtp_rtcp/test/testAPI/test_api.cc

@@ -58,9 +58,9 @@ TEST_F(RtpRtcpAPITest, Basic) {
   EXPECT_EQ(0, module->SetStartTimestamp(test_timestamp));
   EXPECT_EQ(test_timestamp, module->StartTimestamp());
 
-  EXPECT_EQ(false, module->Sending());
+  EXPECT_FALSE(module->Sending());
   EXPECT_EQ(0, module->SetSendingStatus(true));
-  EXPECT_EQ(true, module->Sending());
+  EXPECT_TRUE(module->Sending());
 }
 
 TEST_F(RtpRtcpAPITest, MTU) {
@@ -108,11 +108,11 @@ TEST_F(RtpRtcpAPITest, RTCP) {
   EXPECT_STRCASEEQ(cName, "john.doe@test.test");
   EXPECT_EQ(-1, module->CNAME(NULL));
 
-  EXPECT_EQ(false, module->TMMBR());
+  EXPECT_FALSE(module->TMMBR());
   EXPECT_EQ(0, module->SetTMMBRStatus(true));
-  EXPECT_EQ(true, module->TMMBR());
+  EXPECT_TRUE(module->TMMBR());
   EXPECT_EQ(0, module->SetTMMBRStatus(false));
-  EXPECT_EQ(false, module->TMMBR());
+  EXPECT_FALSE(module->TMMBR());
 
   EXPECT_EQ(kNackOff, module->NACK());
   EXPECT_EQ(0, module->SetNACKStatus(kNackRtcp));

src/modules/rtp_rtcp/test/testAPI/test_api_audio.cc

@@ -65,7 +65,7 @@ class VerifyingAudioReceiver : public RtpData {
           return 0;
         }
       }
-      EXPECT_EQ(false, true) << "This code path should never happen.";
+      ADD_FAILURE() << "This code path should never happen.";
       return -1;
     }
     return 0;
@@ -192,7 +192,7 @@ TEST_F(RtpRtcpAudioTest, Basic) {
   EXPECT_EQ(0, module1->SetSSRC(test_ssrc));
   EXPECT_EQ(0, module1->SetStartTimestamp(test_timestamp));
 
-  EXPECT_EQ(false, module1->TelephoneEvent());
+  EXPECT_FALSE(module1->TelephoneEvent());
 
   // Test detection at the end of a DTMF tone.
   EXPECT_EQ(0, module2->SetTelephoneEventStatus(true, true, true));

src/modules/video_coding/main/source/session_info_unittest.cc

@@ -145,7 +145,7 @@ TEST_F(TestSessionInfo, TestSimpleAPIs) {
   FillPacket(0);
   ASSERT_EQ(kPacketBufferSize,
             session_.InsertPacket(packet_, frame_buffer_, false, 0));
-  EXPECT_EQ(false, session_.HaveLastPacket());
+  EXPECT_FALSE(session_.HaveLastPacket());
   EXPECT_EQ(kVideoFrameKey, session_.FrameType());
 
   packet_.isFirstPacket = false;
@@ -153,7 +153,7 @@ TEST_F(TestSessionInfo, TestSimpleAPIs) {
   packet_.seqNum += 1;
   ASSERT_EQ(kPacketBufferSize,
             session_.InsertPacket(packet_, frame_buffer_, false, 0));
-  EXPECT_EQ(true, session_.HaveLastPacket());
+  EXPECT_TRUE(session_.HaveLastPacket());
   EXPECT_EQ(packet_.seqNum, session_.HighSequenceNumber());
   EXPECT_EQ(0xFFFE, session_.LowSequenceNumber());
 
@@ -804,7 +804,7 @@ TEST_F(TestNackList, NoLosses) {
   EXPECT_EQ(10 * kPacketBufferSize, session_.SessionLength());
   BuildSeqNumList(low, packet_.seqNum);
   EXPECT_EQ(0, session_.BuildHardNackList(seq_num_list_, seq_num_list_length_));
-  EXPECT_EQ(false, session_.session_nack());
+  EXPECT_FALSE(session_.session_nack());
   SCOPED_TRACE("Calling VerifyAll");
   VerifyAll(-1);
 

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

@@ -275,7 +275,6 @@ GenericCodecTest::Perform(CmdArgs& args)
     const float nBitrates = sizeof(bitRate)/sizeof(*bitRate);
     float _bitRate = 0;
     int _frameCnt = 0;
-    WebRtc_Word64 startTime, currentTime, oneSecTime;
     float totalBytesOneSec;//, totalBytesTenSec;
     float totalBytes, actualBitrate;
     VCMFrameCount frameCount; // testing frame type counters
@@ -309,7 +308,6 @@ GenericCodecTest::Perform(CmdArgs& args)
             _vcm->SetChannelParameters((WebRtc_UWord32)_bitRate, 0, 20);
             _frameCnt = 0;
             totalBytes = 0;
-            startTime = _clock->MicrosecondTimestamp();
             _encodeCompleteCallback->Initialize();
             sendStats.SetTargetFrameRate(static_cast<WebRtc_UWord32>(_frameRate));
             _vcm->RegisterSendStatisticsCallback(&sendStats);
@@ -333,7 +331,6 @@ GenericCodecTest::Perform(CmdArgs& args)
                 //currentTime = VCMTickTime::MillisecondTimestamp();//clock()/(double)CLOCKS_PER_SEC;
                 if (_frameCnt == _frameRate)// @ 1sec
                 {
-                    oneSecTime = _clock->MicrosecondTimestamp();
                     totalBytesOneSec =  _encodeCompleteCallback->EncodedBytes();//totalBytes;
                 }
                 TEST(_vcm->TimeUntilNextProcess() >= 0);
@@ -343,7 +340,6 @@ GenericCodecTest::Perform(CmdArgs& args)
             // estimating rates
             // complete sequence
             // bit rate assumes input frame rate is as specified
-            currentTime = _clock->MicrosecondTimestamp();
             totalBytes = _encodeCompleteCallback->EncodedBytes();
             actualBitrate = (float)(8.0/1000)*(totalBytes / (_frameCnt / _frameRate));
 
@@ -367,7 +363,6 @@ GenericCodecTest::Perform(CmdArgs& args)
     /********************************/
     /* Encoder Pipeline Delay Test */
     /******************************/
-    WebRtc_Word32 retVal;
     _vcm->InitializeSender();
     sourceFrame.Free();
     sourceFrame.VerifyAndAllocate(_lengthSourceFrame);
@@ -376,11 +371,11 @@ GenericCodecTest::Perform(CmdArgs& args)
     // going over all available codecs
     for (int k = 0; k < NumberOfCodecs; k++)
     {
-        retVal = _vcm->Codec(k, &_sendCodec);
+        _vcm->Codec(k, &_sendCodec);
-        retVal = _vcm->InitializeSender();
+        _vcm->InitializeSender();
         _sendCodec.maxBitrate = 8000;
-        retVal = _vcm->RegisterSendCodec(&_sendCodec, 4, 1440);
+        _vcm->RegisterSendCodec(&_sendCodec, 4, 1440);
-        retVal = _vcm->RegisterTransportCallback(_encodeCompleteCallback);
+        _vcm->RegisterTransportCallback(_encodeCompleteCallback);
 
         _frameCnt = 0;
         encodeComplete = false;
@@ -393,7 +388,7 @@ GenericCodecTest::Perform(CmdArgs& args)
             sourceFrame.SetWidth(_width);
             _timeStamp += (WebRtc_UWord32)(9e4 / static_cast<float>(_frameRate));
             sourceFrame.SetTimeStamp(_timeStamp);
-            retVal = _vcm->AddVideoFrame(sourceFrame);
+            _vcm->AddVideoFrame(sourceFrame);
             encodeComplete = _encodeCompleteCallback->EncodeComplete();
         } // first frame encoded
         printf ("\n Codec type = %s \n", _sendCodec.plName);

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

@@ -470,40 +470,6 @@ VCMQMDecodeCompleCallback::SetOriginalFrameDimensions(WebRtc_Word32 width, WebRt
 WebRtc_Word32
 VCMQMDecodeCompleCallback::buildInterpolator()
 {
-//    if (_interpolator != NULL)
-//    {
-//        deleteInterpolator(_interpolator);
-//        _interpolator = NULL;
-//    }
-
-    // create decimator
-    WebRtc_Word32 filterPar = 4; //Lanczos (assuming sampling ratio is 1, 1.5, 2, 4)
-
-    float HeightRatio = 0;
-    float WidthRatio = 0;
-
-    WidthRatio = (float)_origWidth/(float)_decWidth;
-    HeightRatio = (float)_origHeight/(float)_decHeight;
-
-    if ( (HeightRatio == 1.0 || HeightRatio == 1.5 || HeightRatio == 2 || HeightRatio == 4) &&
-         (WidthRatio == 1.0 || WidthRatio == 1.5 || WidthRatio == 2 || WidthRatio == 4))
-    {
-        filterPar = 4; //Lanczos
-    } else
-    {
-        filterPar = 0; //BiCubic
-    }
-
-
-    // define interpolator here
-
-    // create interpolator here
-
-//    if (_interpolator == NULL)
-//    {
-//        return -1;
-//    }
-
     WebRtc_UWord32 decFrameLength  = _origWidth*_origHeight*3 >> 1;
     if (_decBuffer != NULL)
     {

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

@@ -67,7 +67,6 @@ int ReceiverTimingTests(CmdArgs& args)
     WebRtc_UWord32 waitTime = 0;
     WebRtc_UWord32 jitterDelayMs = 0;
     WebRtc_UWord32 maxDecodeTimeMs = 0;
-    WebRtc_UWord32 extraDelayMs = 0;
     WebRtc_UWord32 timeStamp = 0;
 
     timing.Reset(static_cast<WebRtc_Word64>(clockInMs + 0.5));
@@ -170,7 +169,6 @@ int ReceiverTimingTests(CmdArgs& args)
 
     jitterDelayMs = 60;
     maxDecodeTimeMs = 10;
-    extraDelayMs = 0;
 
     timeStamp = static_cast<WebRtc_UWord32>(-10000); // To produce a wrap
     clockInMs = 10000.0f;

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

@@ -204,9 +204,9 @@ TEST_F(VideoProcessingModuleTest, FrameStats)
   ASSERT_EQ(_frameLength, fread(_videoFrame.Buffer(), 1, _frameLength,
                                 _sourceFile));
 
-  EXPECT_EQ(false, _vpm->ValidFrameStats(stats));
+  EXPECT_FALSE(_vpm->ValidFrameStats(stats));
   EXPECT_EQ(0, _vpm->GetFrameStats(stats, _videoFrame));
-  EXPECT_EQ(true, _vpm->ValidFrameStats(stats));
+  EXPECT_TRUE(_vpm->ValidFrameStats(stats));
 
   printf("\nFrameStats\n");
   printf("mean: %u\nnumPixels: %u\nsubSamplWidth: "
@@ -218,7 +218,7 @@ TEST_F(VideoProcessingModuleTest, FrameStats)
          static_cast<unsigned int>(stats.sum));
 
   _vpm->ClearFrameStats(stats);
-  EXPECT_EQ(false, _vpm->ValidFrameStats(stats));
+  EXPECT_FALSE(_vpm->ValidFrameStats(stats));
 }
 
 TEST_F(VideoProcessingModuleTest, PreprocessorLogic)

src/modules/video_render/main/test/testAPI/testAPI.cc

@@ -336,7 +336,12 @@ int TestSingleStream(VideoRender* renderModule) {
 
     printf("Start render\n");
     error = renderModule->StartRender(streamId0);
-    assert(error == 0);
+    if (error != 0) {
+      // TODO(phoglund): This test will not work if compiled in release mode.
+      // This rather silly construct here is to avoid compilation errors when
+      // compiling in release. Release => no asserts => unused 'error' variable.
+      assert(false);
+    }
 
     // Loop through an I420 file and render each frame
     const WebRtc_UWord32 width = 352;