diff --git a/src/modules/audio_coding/main/source/acm_codec_database.cc b/src/modules/audio_coding/main/source/acm_codec_database.cc index f38331bff6..2945eebe6f 100644 --- a/src/modules/audio_coding/main/source/acm_codec_database.cc +++ b/src/modules/audio_coding/main/source/acm_codec_database.cc @@ -8,9 +8,6 @@ * be found in the AUTHORS file in the root of the source tree. */ -// 'conversion' conversion from 'type1' to 'type2', possible loss of data -#pragma warning(disable: 4267) - #include #include "acm_codec_database.h" diff --git a/src/modules/audio_coding/main/source/acm_neteq.cc b/src/modules/audio_coding/main/source/acm_neteq.cc index d829db19b5..8294171f4c 100644 --- a/src/modules/audio_coding/main/source/acm_neteq.cc +++ b/src/modules/audio_coding/main/source/acm_neteq.cc @@ -8,8 +8,6 @@ * be found in the AUTHORS file in the root of the source tree. */ -// 'conversion' conversion from 'type1' to 'type2', possible loss of data -#pragma warning(disable: 4267) #include // malloc diff --git a/src/modules/audio_coding/main/source/audio_coding_module.cc b/src/modules/audio_coding/main/source/audio_coding_module.cc index 0ecad8daa0..97c5af9a9a 100644 --- a/src/modules/audio_coding/main/source/audio_coding_module.cc +++ b/src/modules/audio_coding/main/source/audio_coding_module.cc @@ -8,8 +8,6 @@ * be found in the AUTHORS file in the root of the source tree. */ -// 'conversion' conversion from 'type1' to 'type2', possible loss of data -#pragma warning(disable: 4267) #include "acm_dtmf_detection.h" #include "audio_coding_module.h" diff --git a/src/modules/audio_coding/main/test/APITest.cpp b/src/modules/audio_coding/main/test/APITest.cpp index a729c86316..8f7f84df43 100644 --- a/src/modules/audio_coding/main/test/APITest.cpp +++ b/src/modules/audio_coding/main/test/APITest.cpp @@ -206,7 +206,6 @@ APITest::SetUp() { // test if un-registration works; CodecInst nextCodec; - int currentPayloadType = dummyCodec.pltype; AudioCodingModule::Codec(n + 1, nextCodec); nextCodec.pltype = dummyCodec.pltype; if(!FixedPayloadTypeCodec(nextCodec.plname)) @@ -780,7 +779,7 @@ APITest::CheckVADStatus(char side) bool dtxEnabled; bool vadEnabled; ACMVADMode vadMode; - EventWrapper* myEvent = EventWrapper::Create(); + if(side == 'A') { _acmA->VAD(dtxEnabled, vadEnabled, vadMode); @@ -1268,7 +1267,6 @@ void APITest::TestReceiverVAD(char side) { AudioCodingModule* myACM; - EventWrapper* myEvent = EventWrapper::Create(); WebRtc_UWord64* myReceiveVADActivity; if(side == 'A') @@ -1293,9 +1291,9 @@ APITest::TestReceiverVAD(char side) fprintf(stdout, "----------------------------------\n"); fprintf(stdout, "Status........ %s\n", vadStatus? "ON":"OFF"); fprintf(stdout, "mode.......... %d\n", (int)mode); - fprintf(stdout, "VAD Active.... %llu\n", myReceiveVADActivity[0]); - fprintf(stdout, "VAD Passive... %llu\n", myReceiveVADActivity[1]); - fprintf(stdout, "VAD Unknown... %llu\n", myReceiveVADActivity[2]); + fprintf(stdout, "VAD Active.... %lu\n", myReceiveVADActivity[0]); + fprintf(stdout, "VAD Passive... %lu\n", myReceiveVADActivity[1]); + fprintf(stdout, "VAD Unknown... %lu\n", myReceiveVADActivity[2]); } if(vadStatus) @@ -1442,7 +1440,6 @@ void APITest::CurrentCodec(char side) { CodecInst myCodec; - EventWrapper* myEvent = EventWrapper::Create(); if(side == 'A') { _acmA->SendCodec(myCodec); @@ -1478,7 +1475,6 @@ APITest::ChangeCodec(char side) bool* dtx; ACMVADMode* mode; Channel* myChannel; - EventWrapper* myEvent = EventWrapper::Create(); // Reset and Wait if(!_randomTest) { diff --git a/src/modules/audio_coding/main/test/Channel.cpp b/src/modules/audio_coding/main/test/Channel.cpp index bf440eaa03..38493676e6 100644 --- a/src/modules/audio_coding/main/test/Channel.cpp +++ b/src/modules/audio_coding/main/test/Channel.cpp @@ -249,9 +249,9 @@ _saveBitStream(false), _lastPayloadType(-1), _isStereo(false), _leftChannel(true), -_useFECTestWithPacketLoss(false), -_packetLoss(0), _lastInTimestamp(0), +_packetLoss(0), +_useFECTestWithPacketLoss(false), _chID(chID), _beginTime(TickTime::MillisecondTimestamp()), _totalBytes(0) diff --git a/src/modules/audio_coding/main/test/Channel.h b/src/modules/audio_coding/main/test/Channel.h index 396fadcae7..0846788ae2 100644 --- a/src/modules/audio_coding/main/test/Channel.h +++ b/src/modules/audio_coding/main/test/Channel.h @@ -29,17 +29,17 @@ struct ACMTestFrameSizeStats WebRtc_UWord32 numPackets; WebRtc_UWord64 totalPayloadLenByte; WebRtc_UWord64 totalEncodedSamples; - double rateBitPerSec; - double usageLenSec; + double rateBitPerSec; + double usageLenSec; }; struct ACMTestPayloadStats { bool newPacket; - WebRtc_Word16 payloadType; - WebRtc_Word16 lastPayloadLenByte; - WebRtc_UWord32 lastTimestamp; + WebRtc_Word16 payloadType; + WebRtc_Word16 lastPayloadLenByte; + WebRtc_UWord32 lastTimestamp; ACMTestFrameSizeStats frameSizeStats[MAX_NUM_FRAMESIZES]; }; @@ -67,7 +67,7 @@ public: void ResetStats(); WebRtc_Word16 Stats( - CodecInst& codecInst, + CodecInst& codecInst, ACMTestPayloadStats& payloadStats); void Stats( @@ -99,26 +99,26 @@ private: WebRtcRTPHeader& rtpInfo, WebRtc_UWord16 payloadSize); - AudioCodingModule* _receiverACM; - WebRtc_UWord16 _seqNo; - // 60 msec * 32 sample (max) / msec * 2 description (maybe) * 2 bytes / sample - WebRtc_UWord8 _payloadData[60 * 32 * 2 * 2]; + AudioCodingModule* _receiverACM; + WebRtc_UWord16 _seqNo; + // 60msec * 32 sample(max)/msec * 2 description (maybe) * 2 bytes/sample + WebRtc_UWord8 _payloadData[60 * 32 * 2 * 2]; CriticalSectionWrapper* _channelCritSect; - FILE* _bitStreamFile; - bool _saveBitStream; - WebRtc_Word16 _lastPayloadType; - ACMTestPayloadStats _payloadStats[MAX_NUM_PAYLOADS]; - bool _isStereo; - WebRtcRTPHeader _rtpInfo; - bool _leftChannel; - WebRtc_UWord32 _lastInTimestamp; + FILE* _bitStreamFile; + bool _saveBitStream; + WebRtc_Word16 _lastPayloadType; + ACMTestPayloadStats _payloadStats[MAX_NUM_PAYLOADS]; + bool _isStereo; + WebRtcRTPHeader _rtpInfo; + bool _leftChannel; + WebRtc_UWord32 _lastInTimestamp; // FEC Test variables - WebRtc_Word16 _packetLoss; - bool _useFECTestWithPacketLoss; - WebRtc_Word16 _chID; - WebRtc_UWord64 _beginTime; - WebRtc_UWord64 _totalBytes; + WebRtc_Word16 _packetLoss; + bool _useFECTestWithPacketLoss; + WebRtc_Word16 _chID; + WebRtc_UWord64 _beginTime; + WebRtc_UWord64 _totalBytes; }; diff --git a/src/modules/audio_coding/main/test/PCMFile.h b/src/modules/audio_coding/main/test/PCMFile.h index dd8d06f431..dda02b705a 100644 --- a/src/modules/audio_coding/main/test/PCMFile.h +++ b/src/modules/audio_coding/main/test/PCMFile.h @@ -16,22 +16,9 @@ #include #include -// class PCMStream -// { -// protected: -// PCMStream(){} -// ~PCMStream(){} -// public: -// virtual WebRtc_Word32 Read10MsData(AudioFrame& audioFrame) = 0; -// virtual void Write10MsData(WebRtc_Word16 *playoutBuffer, WebRtc_UWord16 playoutLengthSmpls) = 0; -// virtual WebRtc_UWord16 PayloadLength10Ms() const = 0; -// virtual WebRtc_Word32 SamplingFrequency() const = 0; -// }; - - using namespace webrtc; -class PCMFile /*: public PCMStream*/ +class PCMFile { public: PCMFile(); @@ -63,14 +50,14 @@ public: bool readStereo = true); private: FILE* _pcmFile; - WebRtc_UWord16 _nSamples10Ms; - WebRtc_Word32 _frequency; + WebRtc_UWord16 _nSamples10Ms; + WebRtc_Word32 _frequency; bool _endOfFile; bool _autoRewind; bool _rewinded; - WebRtc_UWord32 _timestamp; - bool _saveStereo; + WebRtc_UWord32 _timestamp; bool _readStereo; + bool _saveStereo; }; #endif diff --git a/src/modules/audio_coding/main/test/TestAllCodecs.cpp b/src/modules/audio_coding/main/test/TestAllCodecs.cpp index 15f0f1684d..5f2c9d5edd 100644 --- a/src/modules/audio_coding/main/test/TestAllCodecs.cpp +++ b/src/modules/audio_coding/main/test/TestAllCodecs.cpp @@ -780,8 +780,6 @@ void TestAllCodecs::Run(TestPack* channel) { AudioFrame audioFrame; - WebRtc_UWord16 SamplesIn10MsecA = _inFileA.PayloadLength10Ms(); - WebRtc_UWord32 timestampA = 1; WebRtc_Word32 outFreqHzB = _outFileB.SamplingFrequency(); WebRtc_UWord16 recSize; WebRtc_UWord32 timeStampDiff; diff --git a/src/modules/audio_coding/main/test/TestFEC.cpp b/src/modules/audio_coding/main/test/TestFEC.cpp index 30bb25ee69..ed61828106 100644 --- a/src/modules/audio_coding/main/test/TestFEC.cpp +++ b/src/modules/audio_coding/main/test/TestFEC.cpp @@ -566,19 +566,13 @@ void TestFEC::Run() WebRtc_UWord16 msecPassed = 0; WebRtc_UWord32 secPassed = 0; - WebRtc_UWord16 SamplesIn10MsecA = _inFileA.PayloadLength10Ms(); - WebRtc_UWord32 timestampA = 1; WebRtc_Word32 outFreqHzB = _outFileB.SamplingFrequency(); while(!_inFileA.EndOfFile()) { _inFileA.Read10MsData(audioFrame); - //audioFrame._timeStamp = timestampA; - //timestampA += SamplesIn10MsecA; CHECK_ERROR(_acmA->Add10MsData(audioFrame)); - CHECK_ERROR(_acmA->Process()); - CHECK_ERROR(_acmB->PlayoutData10Ms(outFreqHzB, audioFrame)); _outFileB.Write10MsData(audioFrame._payloadData, audioFrame._payloadDataLengthInSamples); msecPassed += 10; diff --git a/src/modules/audio_coding/main/test/TestStereo.cpp b/src/modules/audio_coding/main/test/TestStereo.cpp index 1add2409de..a8618f08ad 100644 --- a/src/modules/audio_coding/main/test/TestStereo.cpp +++ b/src/modules/audio_coding/main/test/TestStereo.cpp @@ -56,10 +56,7 @@ TestPackStereo::SendData( WebRtc_Word32 status; WebRtc_UWord16 payloadDataSize = payloadSize; WebRtc_UWord8 payloadDataMaster[60 * 32 * 2 * 2]; - WebRtc_UWord8 payloadDataSlave[60 * 32 * 2 * 2]; - bool twoBytePerSample = false; - bool oneBytePerSample = true; - bool frameBased = false; + WebRtc_UWord8 payloadDataSlave[60 * 32 * 2 * 2]; rtpInfo.header.markerBit = false; rtpInfo.header.ssrc = 0; @@ -470,8 +467,6 @@ void TestStereo::Run(TestPackStereo* channel) { AudioFrame audioFrame; - WebRtc_UWord16 SamplesIn10MsecA = _inFileA.PayloadLength10Ms(); - WebRtc_UWord32 timestampA = 1; WebRtc_Word32 outFreqHzB = _outFileB.SamplingFrequency(); WebRtc_UWord16 recSize; WebRtc_UWord32 timeStampDiff; diff --git a/src/modules/audio_coding/main/test/TestVADDTX.cpp b/src/modules/audio_coding/main/test/TestVADDTX.cpp index 768bdde034..3801653eb6 100644 --- a/src/modules/audio_coding/main/test/TestVADDTX.cpp +++ b/src/modules/audio_coding/main/test/TestVADDTX.cpp @@ -92,8 +92,6 @@ void TestVADDTX::Perform() WebRtc_Word16 testCntr = 1; - VADDTXstruct setDTX, getDTX, expectedDTX; - bool dtxReplaced; WebRtc_Word16 testResults = 0; #ifdef WEBRTC_CODEC_ISAC @@ -144,7 +142,7 @@ void TestVADDTX::Perform() printf("VAD/DTX test completed with %d subtests failed\n", testResults); if (testResults > 0) { - printf("Press return\n\n", testResults); + printf("Press return\n\n"); getchar(); } } @@ -227,12 +225,13 @@ void TestVADDTX::SetVAD(bool statusDTX, bool statusVAD, WebRtc_Word16 vadMode) printf("DTX: %s not the same as requested: %s\n", dtxEnabled? "ON":"OFF", dtxEnabled? "OFF":"ON"); } - if((statusVAD == true) && (vadEnabled == false) || - (statusVAD == false) && (vadEnabled == false) && (statusDTX == true)) + if(((statusVAD == true) && (vadEnabled == false)) || + ((statusVAD == false) && (vadEnabled == false) && + (statusDTX == true))) { printf("VAD: %s not the same as requested: %s\n", vadEnabled? "ON":"OFF", vadEnabled? "OFF":"ON"); - } + } if(vadModeSet != vadMode) { printf("VAD mode: %d not the same as requested: %d\n", diff --git a/src/modules/audio_coding/main/test/utility.cpp b/src/modules/audio_coding/main/test/utility.cpp index c654019219..58a2cf13f4 100644 --- a/src/modules/audio_coding/main/test/utility.cpp +++ b/src/modules/audio_coding/main/test/utility.cpp @@ -158,13 +158,13 @@ PrintCodecs() } CircularBuffer::CircularBuffer(WebRtc_UWord32 len): +_buff(NULL), +_idx(0), _buffIsFull(false), _calcAvg(false), _calcVar(false), _sum(0), -_sumSqr(0), -_idx(0), -_buff(NULL) +_sumSqr(0) { _buff = new(double[len]); if(_buff == NULL)