diff --git a/src/modules/audio_coding/main/test/APITest.cc b/src/modules/audio_coding/main/test/APITest.cc index 07bbd5252f..a634f4aa1d 100644 --- a/src/modules/audio_coding/main/test/APITest.cc +++ b/src/modules/audio_coding/main/test/APITest.cc @@ -81,8 +81,6 @@ _processEventB(NULL), _apiEventB(NULL), _codecCntrA(0), _codecCntrB(0), -_testCntrA(1), -_testCntrB(1), _thereIsEncoderA(false), _thereIsEncoderB(false), _thereIsDecoderA(false), @@ -118,7 +116,7 @@ _testNumB(1) _receiveVADActivityA[n] = 0; _receiveVADActivityB[n] = 0; } - + _movingDot[40] = '\0'; for(int n = 0; n <40; n++) @@ -172,7 +170,7 @@ APITest::~APITest() // return _outFile.Open(fileName, frequencyHz, "wb"); //} -WebRtc_Word16 +WebRtc_Word16 APITest::SetUp() { _acmA = AudioCodingModule::Create(1); @@ -260,7 +258,7 @@ APITest::SetUp() char fileName[500]; WebRtc_UWord16 frequencyHz; - + printf("\n\nAPI Test\n"); printf("========\n"); printf("Hit enter to accept the default values indicated in []\n\n"); @@ -301,7 +299,7 @@ APITest::SetUp() _channel_B2A = new Channel(1); CHECK_ERROR_MT(_acmB->RegisterTransportCallback(_channel_B2A)); _channel_B2A->RegisterReceiverACM(_acmA); - + //--- EVENT TIMERS // A _pullEventA = EventWrapper::Create(); @@ -321,7 +319,7 @@ APITest::SetUp() _outFreqHzB = _outFileB.SamplingFrequency(); - //Trace::SetEncryptedTraceFile("ACMAPITestEncrypted.txt"); + //Trace::SetEncryptedTraceFile("ACMAPITestEncrypted.txt"); char print[11]; @@ -357,29 +355,29 @@ APITest::SetUp() #endif _vadCallbackA = new VADCallback; _vadCallbackB = new VADCallback; - + return 0; } -bool +bool APITest::PushAudioThreadA(void* obj) { return static_cast(obj)->PushAudioRunA(); } -bool +bool APITest::PushAudioThreadB(void* obj) { return static_cast(obj)->PushAudioRunB(); } -bool +bool APITest::PullAudioThreadA(void* obj) { return static_cast(obj)->PullAudioRunA(); } -bool +bool APITest::PullAudioThreadB(void* obj) { return static_cast(obj)->PullAudioRunB(); @@ -409,7 +407,7 @@ APITest::APIThreadB(void* obj) return static_cast(obj)->APIRunB(); } -bool +bool APITest::PullAudioRunA() { _pullEventA->Wait(100); @@ -437,7 +435,7 @@ APITest::PullAudioRunA() return true; } -bool +bool APITest::PullAudioRunB() { _pullEventB->Wait(100); @@ -462,11 +460,11 @@ APITest::PullAudioRunB() _outFileB.Write10MsData(audioFrame); } _receiveVADActivityB[(int)audioFrame.vad_activity_]++; - } + } return true; } -bool +bool APITest::PushAudioRunA() { _pushEventA->Wait(100); @@ -487,7 +485,7 @@ APITest::PushAudioRunA() return true; } -bool +bool APITest::PushAudioRunB() { _pushEventB->Wait(100); @@ -567,7 +565,7 @@ APITest::RunTest(char thread) { _testNumA = (_testNumB + 1 + (rand() % 6)) % 7; testNum = _testNumA; - + _movingDot[_dotPositionA] = ' '; if(_dotPositionA == 0) { @@ -577,7 +575,7 @@ APITest::RunTest(char thread) { _dotMoveDirectionA = -1; } - _dotPositionA += _dotMoveDirectionA; + _dotPositionA += _dotMoveDirectionA; _movingDot[_dotPositionA] = (_dotMoveDirectionA > 0)? '>':'<'; } else @@ -594,7 +592,7 @@ APITest::RunTest(char thread) { _dotMoveDirectionB = -1; } - _dotPositionB += _dotMoveDirectionB; + _dotPositionB += _dotMoveDirectionB; _movingDot[_dotPositionB] = (_dotMoveDirectionB > 0)? '>':'<'; } //fprintf(stderr, "%c: %d \n", thread, testNum); @@ -617,7 +615,7 @@ APITest::RunTest(char thread) TestDelay('A'); break; case 3: - TestSendVAD('A'); + TestSendVAD('A'); break; case 4: TestRegisteration('A'); @@ -641,7 +639,7 @@ APITest::RunTest(char thread) bool APITest::APIRunA() -{ +{ _apiEventA->Wait(50); bool randomTest; @@ -664,7 +662,7 @@ APITest::APIRunA() TestDelay('A'); } // VAD TEST - TestSendVAD('A'); + TestSendVAD('A'); TestRegisteration('A'); TestReceiverVAD('A'); #ifdef WEBRTC_DTMF_DETECTION @@ -676,7 +674,7 @@ APITest::APIRunA() bool APITest::APIRunB() -{ +{ _apiEventB->Wait(50); bool randomTest; { @@ -688,7 +686,7 @@ APITest::APIRunB() { RunTest('B'); } - + return true; } @@ -700,46 +698,46 @@ APITest::Perform() //--- THREADS // A // PUSH - ThreadWrapper* myPushAudioThreadA = ThreadWrapper::CreateThread(PushAudioThreadA, + ThreadWrapper* myPushAudioThreadA = ThreadWrapper::CreateThread(PushAudioThreadA, this, kNormalPriority, "PushAudioThreadA"); CHECK_THREAD_NULLITY(myPushAudioThreadA, "Unable to start A::PUSH thread"); // PULL - ThreadWrapper* myPullAudioThreadA = ThreadWrapper::CreateThread(PullAudioThreadA, + ThreadWrapper* myPullAudioThreadA = ThreadWrapper::CreateThread(PullAudioThreadA, this, kNormalPriority, "PullAudioThreadA"); CHECK_THREAD_NULLITY(myPullAudioThreadA, "Unable to start A::PULL thread"); // Process - ThreadWrapper* myProcessThreadA = ThreadWrapper::CreateThread(ProcessThreadA, + ThreadWrapper* myProcessThreadA = ThreadWrapper::CreateThread(ProcessThreadA, this, kNormalPriority, "ProcessThreadA"); CHECK_THREAD_NULLITY(myProcessThreadA, "Unable to start A::Process thread"); - // API - ThreadWrapper* myAPIThreadA = ThreadWrapper::CreateThread(APIThreadA, + // API + ThreadWrapper* myAPIThreadA = ThreadWrapper::CreateThread(APIThreadA, this, kNormalPriority, "APIThreadA"); CHECK_THREAD_NULLITY(myAPIThreadA, "Unable to start A::API thread"); // B // PUSH - ThreadWrapper* myPushAudioThreadB = ThreadWrapper::CreateThread(PushAudioThreadB, + ThreadWrapper* myPushAudioThreadB = ThreadWrapper::CreateThread(PushAudioThreadB, this, kNormalPriority, "PushAudioThreadB"); CHECK_THREAD_NULLITY(myPushAudioThreadB, "Unable to start B::PUSH thread"); // PULL - ThreadWrapper* myPullAudioThreadB = ThreadWrapper::CreateThread(PullAudioThreadB, + ThreadWrapper* myPullAudioThreadB = ThreadWrapper::CreateThread(PullAudioThreadB, this, kNormalPriority, "PullAudioThreadB"); CHECK_THREAD_NULLITY(myPullAudioThreadB, "Unable to start B::PULL thread"); // Process - ThreadWrapper* myProcessThreadB = ThreadWrapper::CreateThread(ProcessThreadB, + ThreadWrapper* myProcessThreadB = ThreadWrapper::CreateThread(ProcessThreadB, this, kNormalPriority, "ProcessThreadB"); CHECK_THREAD_NULLITY(myProcessThreadB, "Unable to start B::Process thread"); // API - ThreadWrapper* myAPIThreadB = ThreadWrapper::CreateThread(APIThreadB, + ThreadWrapper* myAPIThreadB = ThreadWrapper::CreateThread(APIThreadB, this, kNormalPriority, "APIThreadB"); CHECK_THREAD_NULLITY(myAPIThreadB, "Unable to start B::API thread"); - + //_apiEventA->StartTimer(true, 5000); //_apiEventB->StartTimer(true, 5000); _processEventA->StartTimer(true, 10); _processEventB->StartTimer(true, 10); - + _pullEventA->StartTimer(true, 10); _pullEventB->StartTimer(true, 10); @@ -764,7 +762,7 @@ APITest::Perform() //completeEvent->Wait(0xFFFFFFFF);//(unsigned long)((unsigned long)TEST_DURATION_SEC * (unsigned long)1000)); delete completeEvent; - + myPushAudioThreadA->Stop(); myPullAudioThreadA->Stop(); myProcessThreadA->Stop(); @@ -802,12 +800,12 @@ APITest::CheckVADStatus(char side) _acmA->RegisterVADCallback(NULL); _vadCallbackA->Reset(); _acmA->RegisterVADCallback(_vadCallbackA); - + if(!_randomTest) { if(_verbose) { - fprintf(stdout, "DTX %3s, VAD %3s, Mode %d", + fprintf(stdout, "DTX %3s, VAD %3s, Mode %d", dtxEnabled? "ON":"OFF", vadEnabled? "ON":"OFF", (int)vadMode); @@ -818,7 +816,7 @@ APITest::CheckVADStatus(char side) else { Wait(5000); - fprintf(stdout, "DTX %3s, VAD %3s, Mode %d => bit-rate %3.0f kbps\n", + fprintf(stdout, "DTX %3s, VAD %3s, Mode %d => bit-rate %3.0f kbps\n", dtxEnabled? "ON":"OFF", vadEnabled? "ON":"OFF", (int)vadMode, @@ -847,12 +845,12 @@ APITest::CheckVADStatus(char side) _acmB->RegisterVADCallback(NULL); _vadCallbackB->Reset(); _acmB->RegisterVADCallback(_vadCallbackB); - + if(!_randomTest) { if(_verbose) { - fprintf(stdout, "DTX %3s, VAD %3s, Mode %d", + fprintf(stdout, "DTX %3s, VAD %3s, Mode %d", dtxEnabled? "ON":"OFF", vadEnabled? "ON":"OFF", (int)vadMode); @@ -863,7 +861,7 @@ APITest::CheckVADStatus(char side) else { Wait(5000); - fprintf(stdout, "DTX %3s, VAD %3s, Mode %d => bit-rate %3.0f kbps\n", + fprintf(stdout, "DTX %3s, VAD %3s, Mode %d => bit-rate %3.0f kbps\n", dtxEnabled? "ON":"OFF", vadEnabled? "ON":"OFF", (int)vadMode, @@ -898,7 +896,7 @@ APITest::TestDelay(char side) WebRtc_UWord32 inTimestamp = 0; WebRtc_UWord32 outTimestamp = 0; - double estimDelay = 0; + double estimDelay = 0; double averageEstimDelay = 0; double averageDelay = 0; @@ -923,7 +921,7 @@ APITest::TestDelay(char side) CHECK_ERROR_MT(myACM->SetMinimumPlayoutDelay(*myMinDelay)); - inTimestamp = myChannel->LastInTimestamp(); + inTimestamp = myChannel->LastInTimestamp(); CHECK_ERROR_MT(myACM->PlayoutTimestamp(outTimestamp)); if(!_randomTest) @@ -935,11 +933,11 @@ APITest::TestDelay(char side) { myEvent->Wait(1000); - inTimestamp = myChannel->LastInTimestamp(); + inTimestamp = myChannel->LastInTimestamp(); CHECK_ERROR_MT(myACM->PlayoutTimestamp(outTimestamp)); //std::cout << outTimestamp << std::endl << std::flush; - estimDelay = (double)((WebRtc_UWord32)(inTimestamp - outTimestamp)) / + estimDelay = (double)((WebRtc_UWord32)(inTimestamp - outTimestamp)) / ((double)myACM->ReceiveFrequency() / 1000.0); estimDelayCB.Update(estimDelay); @@ -970,7 +968,7 @@ APITest::TestDelay(char side) } *myMinDelay = (rand() % 1000) + 1; - + ACMNetworkStatistics networkStat; CHECK_ERROR_MT(myACM->NetworkStatistics(networkStat)); @@ -978,12 +976,12 @@ APITest::TestDelay(char side) { fprintf(stdout, "\n\nJitter Statistics at Side %c\n", side); fprintf(stdout, "--------------------------------------\n"); - fprintf(stdout, "buffer-size............. %d\n", networkStat.currentBufferSize); + fprintf(stdout, "buffer-size............. %d\n", networkStat.currentBufferSize); fprintf(stdout, "Preferred buffer-size... %d\n", networkStat.preferredBufferSize); fprintf(stdout, "Peaky jitter mode........%d\n", networkStat.jitterPeaksFound); fprintf(stdout, "packet-size rate........ %d\n", networkStat.currentPacketLossRate); - fprintf(stdout, "discard rate............ %d\n", networkStat.currentDiscardRate); - fprintf(stdout, "expand rate............. %d\n", networkStat.currentExpandRate); + fprintf(stdout, "discard rate............ %d\n", networkStat.currentDiscardRate); + fprintf(stdout, "expand rate............. %d\n", networkStat.currentExpandRate); fprintf(stdout, "Preemptive rate......... %d\n", networkStat.currentPreemptiveRate); fprintf(stdout, "Accelerate rate......... %d\n", networkStat.currentAccelerateRate); fprintf(stdout, "Clock-drift............. %d\n", networkStat.clockDriftPPM); @@ -1020,7 +1018,7 @@ APITest::TestRegisteration(char sendSide) fprintf(stdout, " Unregister/register Receive Codec\n"); fprintf(stdout, "---------------------------------------------------------\n"); } - + switch(sendSide) { case 'A': @@ -1179,7 +1177,7 @@ APITest::TestPlayout(char receiveSide) CHECK_ERROR_MT(receiveFreqHz); CHECK_ERROR_MT(playoutFreqHz); - + char bgnString[25]; switch(*bgnMode) { @@ -1400,7 +1398,7 @@ APITest::TestSendVAD(char side) // Fault Test CHECK_PROTECTED_MT(myACM->SetVAD(false, true, (ACMVADMode)-1)); CHECK_PROTECTED_MT(myACM->SetVAD(false, true, (ACMVADMode)4)); - + } @@ -1477,14 +1475,14 @@ APITest::ChangeCodec(char side) myChannel = _channel_B2A; } - myACM->ResetEncoder(); + myACM->ResetEncoder(); Wait(100); // Register the next codec do { - *codecCntr = (*codecCntr < AudioCodingModule::NumberOfCodecs() - 1)? - (*codecCntr + 1):0; + *codecCntr = (*codecCntr < AudioCodingModule::NumberOfCodecs() - 1)? + (*codecCntr + 1):0; if(*codecCntr == 0) { @@ -1494,7 +1492,7 @@ APITest::ChangeCodec(char side) *thereIsEncoder = false; } CHECK_ERROR_MT(myACM->InitializeSender()); - Wait(1000); + Wait(1000); // After Initialization CN is lost, re-register them if(AudioCodingModule::Codec("CN", myCodec, 8000, 1) >= 0) @@ -1541,8 +1539,8 @@ APITest::ChangeCodec(char side) Wait(500); } - -void + +void APITest::LookForDTMF(char side) { if(!_randomTest) @@ -1550,11 +1548,11 @@ APITest::LookForDTMF(char side) fprintf(stdout, "\n\nLooking for DTMF Signal in Side %c\n", side); fprintf(stdout, "----------------------------------------\n"); } - + if(side == 'A') { _acmB->RegisterIncomingMessagesCallback(NULL); - _acmA->RegisterIncomingMessagesCallback(_dtmfCallback); + _acmA->RegisterIncomingMessagesCallback(_dtmfCallback); Wait(1000); _acmA->RegisterIncomingMessagesCallback(NULL); } diff --git a/src/modules/audio_coding/main/test/APITest.h b/src/modules/audio_coding/main/test/APITest.h index db0a87caf7..ee3f5e6c99 100644 --- a/src/modules/audio_coding/main/test/APITest.h +++ b/src/modules/audio_coding/main/test/APITest.h @@ -1,5 +1,5 @@ /* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. * * Use of this source code is governed by a BSD-style license * that can be found in the LICENSE file in the root of the source @@ -30,7 +30,7 @@ public: void Perform(); private: WebRtc_Word16 SetUp(); - + static bool PushAudioThreadA(void* obj); static bool PullAudioThreadA(void* obj); static bool ProcessThreadA(void* obj); @@ -56,24 +56,24 @@ private: // set/get receiver VAD status & mode. void TestReceiverVAD(char side); - // + // void TestSendVAD(char side); void CurrentCodec(char side); - + void ChangeCodec(char side); - + void Wait(WebRtc_UWord32 waitLengthMs); void LookForDTMF(char side); void RunTest(char thread); - - bool PushAudioRunA(); + + bool PushAudioRunA(); bool PullAudioRunA(); bool ProcessRunA(); bool APIRunA(); - + bool PullAudioRunB(); bool PushAudioRunB(); bool ProcessRunB(); @@ -84,11 +84,11 @@ private: //--- ACMs AudioCodingModule* _acmA; AudioCodingModule* _acmB; - + //--- Channels Channel* _channel_A2B; Channel* _channel_B2A; - + //--- I/O files // A PCMFile _inFileA; @@ -96,13 +96,13 @@ private: // B PCMFile _outFileB; PCMFile _inFileB; - + //--- I/O params // A WebRtc_Word32 _outFreqHzA; // B WebRtc_Word32 _outFreqHzB; - + // Should we write to file. // we might skip writing to file if we // run the test for a long time. @@ -123,10 +123,6 @@ private: WebRtc_UWord8 _codecCntrA; WebRtc_UWord8 _codecCntrB; - // keep track of tests - WebRtc_UWord8 _testCntrA; - WebRtc_UWord8 _testCntrB; - // Is set to true if there is no encoder in either side bool _thereIsEncoderA; bool _thereIsEncoderB; @@ -144,7 +140,7 @@ private: WebRtc_Word32 _minDelayA; WebRtc_Word32 _minDelayB; bool _payloadUsed[32]; - + AudioPlayoutMode _playoutModeA; AudioPlayoutMode _playoutModeB; @@ -155,14 +151,14 @@ private: int _receiveVADActivityA[3]; int _receiveVADActivityB[3]; bool _verbose; - + int _dotPositionA; int _dotMoveDirectionA; int _dotPositionB; int _dotMoveDirectionB; char _movingDot[41]; - + DTMFDetector* _dtmfCallback; VADCallback* _vadCallbackA; VADCallback* _vadCallbackB; diff --git a/src/modules/audio_coding/main/test/Channel.cc b/src/modules/audio_coding/main/test/Channel.cc index 4b0f7d0091..d7e387aca7 100644 --- a/src/modules/audio_coding/main/test/Channel.cc +++ b/src/modules/audio_coding/main/test/Channel.cc @@ -19,12 +19,12 @@ namespace webrtc { -WebRtc_Word32 +WebRtc_Word32 Channel::SendData( const FrameType frameType, const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp, - const WebRtc_UWord8* payloadData, + const WebRtc_UWord8* payloadData, const WebRtc_UWord16 payloadSize, const RTPFragmentationHeader* fragmentation) { @@ -104,7 +104,7 @@ Channel::SendData( } } } - + _channelCritSect->Enter(); if(_saveBitStream) { @@ -135,9 +135,9 @@ Channel::SendData( return status; } -void +void Channel::CalcStatistics( - WebRtcRTPHeader& rtpInfo, + WebRtcRTPHeader& rtpInfo, WebRtc_UWord16 payloadSize) { int n; @@ -146,7 +146,7 @@ Channel::CalcStatistics( { // payload-type is changed. // we have to terminate the calculations on the previous payload type - // we ignore the last packet in that payload type just to make things + // we ignore the last packet in that payload type just to make things // easier. for(n = 0; n < MAX_NUM_PAYLOADS; n++) { @@ -180,12 +180,12 @@ Channel::CalcStatistics( assert(lastFrameSizeSample > 0); int k = 0; while((currentPayloadStr->frameSizeStats[k].frameSizeSample != - lastFrameSizeSample) && + lastFrameSizeSample) && (currentPayloadStr->frameSizeStats[k].frameSizeSample != 0)) { k++; } - ACMTestFrameSizeStats* currentFrameSizeStats = + ACMTestFrameSizeStats* currentFrameSizeStats = &(currentPayloadStr->frameSizeStats[k]); currentFrameSizeStats->frameSizeSample = (WebRtc_Word16)lastFrameSizeSample; @@ -197,15 +197,15 @@ Channel::CalcStatistics( // increment the total number of bytes (this is based on // the previous payload we don't know the frame-size of // the current payload. - currentFrameSizeStats->totalPayloadLenByte += + currentFrameSizeStats->totalPayloadLenByte += currentPayloadStr->lastPayloadLenByte; // store the maximum payload-size (this is based on // the previous payload we don't know the frame-size of // the current payload. - if(currentFrameSizeStats->maxPayloadLen < + if(currentFrameSizeStats->maxPayloadLen < currentPayloadStr->lastPayloadLenByte) { - currentFrameSizeStats->maxPayloadLen = + currentFrameSizeStats->maxPayloadLen = currentPayloadStr->lastPayloadLenByte; } // store the current values for the next time @@ -247,7 +247,6 @@ _leftChannel(true), _lastInTimestamp(0), _packetLoss(0), _useFECTestWithPacketLoss(false), -_chID(chID), _beginTime(TickTime::MillisecondTimestamp()), _totalBytes(0) { @@ -270,7 +269,7 @@ _totalBytes(0) { _saveBitStream = true; char bitStreamFileName[500]; - sprintf(bitStreamFileName, "bitStream_%d.dat", chID); + sprintf(bitStreamFileName, "bitStream_%d.dat", chID); _bitStreamFile = fopen(bitStreamFileName, "wb"); } else @@ -284,14 +283,14 @@ Channel::~Channel() delete _channelCritSect; } -void +void Channel::RegisterReceiverACM(AudioCodingModule* acm) { _receiverACM = acm; return; } -void +void Channel::ResetStats() { int n; @@ -316,7 +315,7 @@ Channel::ResetStats() _channelCritSect->Leave(); } -WebRtc_Word16 +WebRtc_Word16 Channel::Stats(CodecInst& codecInst, ACMTestPayloadStats& payloadStats) { _channelCritSect->Enter(); @@ -342,12 +341,12 @@ Channel::Stats(CodecInst& codecInst, ACMTestPayloadStats& payloadStats) _channelCritSect->Leave(); return 0; } - payloadStats.frameSizeStats[n].usageLenSec = + payloadStats.frameSizeStats[n].usageLenSec = (double)payloadStats.frameSizeStats[n].totalEncodedSamples / (double)codecInst.plfreq; - payloadStats.frameSizeStats[n].rateBitPerSec = - payloadStats.frameSizeStats[n].totalPayloadLenByte * 8 / + payloadStats.frameSizeStats[n].rateBitPerSec = + payloadStats.frameSizeStats[n].totalPayloadLenByte * 8 / payloadStats.frameSizeStats[n].usageLenSec; } @@ -355,7 +354,7 @@ Channel::Stats(CodecInst& codecInst, ACMTestPayloadStats& payloadStats) return 0; } -void +void Channel::Stats(WebRtc_UWord32* numPackets) { _channelCritSect->Enter(); @@ -375,18 +374,18 @@ Channel::Stats(WebRtc_UWord32* numPackets) { break; } - numPackets[k] += + numPackets[k] += _payloadStats[k].frameSizeStats[n].numPackets; } } _channelCritSect->Leave(); } -void +void Channel::Stats(WebRtc_UWord8* payloadType, WebRtc_UWord32* payloadLenByte) { _channelCritSect->Enter(); - + int k; int n; memset(payloadLenByte, 0, MAX_NUM_PAYLOADS * sizeof(WebRtc_UWord32)); @@ -418,7 +417,7 @@ Channel::PrintStats(CodecInst& codecInst) { ACMTestPayloadStats payloadStats; Stats(codecInst, payloadStats); - printf("%s %d kHz\n", + printf("%s %d kHz\n", codecInst.plname, codecInst.plfreq / 1000); printf("=====================================================\n"); @@ -435,19 +434,19 @@ Channel::PrintStats(CodecInst& codecInst) { break; } - printf("Frame-size.................... %d samples\n", + printf("Frame-size.................... %d samples\n", payloadStats.frameSizeStats[k].frameSizeSample); - printf("Average Rate.................. %.0f bits/sec\n", + printf("Average Rate.................. %.0f bits/sec\n", payloadStats.frameSizeStats[k].rateBitPerSec); printf("Maximum Payload-Size.......... %d Bytes\n", payloadStats.frameSizeStats[k].maxPayloadLen); printf("Maximum Instantaneous Rate.... %.0f bits/sec\n", - ((double)payloadStats.frameSizeStats[k].maxPayloadLen * 8.0 * - (double)codecInst.plfreq) / + ((double)payloadStats.frameSizeStats[k].maxPayloadLen * 8.0 * + (double)codecInst.plfreq) / (double)payloadStats.frameSizeStats[k].frameSizeSample); printf("Number of Packets............. %u\n", (unsigned int)payloadStats.frameSizeStats[k].numPackets); - printf("Duration...................... %0.3f sec\n\n", + printf("Duration...................... %0.3f sec\n\n", payloadStats.frameSizeStats[k].usageLenSec); } @@ -473,6 +472,6 @@ Channel::BitRate() rate = ((double)_totalBytes * 8.0)/ (double)(currTime - _beginTime); _channelCritSect->Leave(); return rate; -} +} } // namespace webrtc diff --git a/src/modules/audio_coding/main/test/Channel.h b/src/modules/audio_coding/main/test/Channel.h index 375bec79d5..617027e7d0 100644 --- a/src/modules/audio_coding/main/test/Channel.h +++ b/src/modules/audio_coding/main/test/Channel.h @@ -1,5 +1,5 @@ /* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. * * Use of this source code is governed by a BSD-style license * that can be found in the LICENSE file in the root of the source @@ -32,7 +32,7 @@ struct ACMTestFrameSizeStats WebRtc_UWord64 totalEncodedSamples; double rateBitPerSec; double usageLenSec; - + }; struct ACMTestPayloadStats @@ -56,36 +56,36 @@ public: const FrameType frameType, const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp, - const WebRtc_UWord8* payloadData, + const WebRtc_UWord8* payloadData, const WebRtc_UWord16 payloadSize, const RTPFragmentationHeader* fragmentation); void RegisterReceiverACM( AudioCodingModule *acm); - + void ResetStats(); - + WebRtc_Word16 Stats( CodecInst& codecInst, ACMTestPayloadStats& payloadStats); - + void Stats( WebRtc_UWord32* numPackets); - + void Stats( - WebRtc_UWord8* payloadLenByte, + WebRtc_UWord8* payloadLenByte, WebRtc_UWord32* payloadType); - + void PrintStats( CodecInst& codecInst); - + void SetIsStereo(bool isStereo) { _isStereo = isStereo; } WebRtc_UWord32 LastInTimestamp(); - + void SetFECTestWithPacketLoss(bool usePacketLoss) { _useFECTestWithPacketLoss = usePacketLoss; @@ -115,7 +115,6 @@ private: // FEC Test variables WebRtc_Word16 _packetLoss; bool _useFECTestWithPacketLoss; - WebRtc_Word16 _chID; WebRtc_UWord64 _beginTime; WebRtc_UWord64 _totalBytes; }; diff --git a/src/modules/audio_coding/main/test/EncodeDecodeTest.cc b/src/modules/audio_coding/main/test/EncodeDecodeTest.cc index 98c01fdf24..8510e00532 100644 --- a/src/modules/audio_coding/main/test/EncodeDecodeTest.cc +++ b/src/modules/audio_coding/main/test/EncodeDecodeTest.cc @@ -50,8 +50,6 @@ Sender::Sender() : _acm(NULL), _pcmFile(), _audioFrame(), - _payloadSize(0), - _timeStamp(0), _packetization(NULL) { } diff --git a/src/modules/audio_coding/main/test/EncodeDecodeTest.h b/src/modules/audio_coding/main/test/EncodeDecodeTest.h index 4b4dba8fc8..f407a6b357 100644 --- a/src/modules/audio_coding/main/test/EncodeDecodeTest.h +++ b/src/modules/audio_coding/main/test/EncodeDecodeTest.h @@ -61,8 +61,6 @@ class Sender { AudioCodingModule* _acm; PCMFile _pcmFile; AudioFrame _audioFrame; - WebRtc_UWord16 _payloadSize; - WebRtc_UWord32 _timeStamp; TestPacketization* _packetization; }; @@ -81,7 +79,6 @@ class Receiver { private: AudioCodingModule* _acm; - bool _rtpEOF; RTPStream* _rtpStream; PCMFile _pcmFile; WebRtc_Word16* _playoutBuffer; @@ -110,7 +107,7 @@ class EncodeDecodeTest: public ACMTest { protected: Sender _sender; Receiver _receiver; -}; +}; } // namespace webrtc diff --git a/src/modules/audio_coding/main/test/TestStereo.h b/src/modules/audio_coding/main/test/TestStereo.h index ed8c1b83df..3023139b5b 100644 --- a/src/modules/audio_coding/main/test/TestStereo.h +++ b/src/modules/audio_coding/main/test/TestStereo.h @@ -48,7 +48,6 @@ class TestPackStereo : public AudioPacketizationCallback { private: AudioCodingModule* receiver_acm_; WebRtc_Word16 seq_no_; - WebRtc_UWord8 payload_data_[60 * 32 * 2 * 2]; WebRtc_UWord32 timestamp_diff_; WebRtc_UWord32 last_in_timestamp_; WebRtc_UWord64 total_bytes_; diff --git a/src/modules/audio_coding/main/test/TwoWayCommunication.h b/src/modules/audio_coding/main/test/TwoWayCommunication.h index 0b3331757c..002244c328 100644 --- a/src/modules/audio_coding/main/test/TwoWayCommunication.h +++ b/src/modules/audio_coding/main/test/TwoWayCommunication.h @@ -1,5 +1,5 @@ /* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. * * Use of this source code is governed by a BSD-style license * that can be found in the LICENSE file in the root of the source @@ -53,9 +53,6 @@ private: PCMFile _outFileRefA; PCMFile _outFileRefB; - DTMFDetector* _dtmfDetectorA; - DTMFDetector* _dtmfDetectorB; - int _testMode; }; diff --git a/src/modules/audio_device/main/source/mac/audio_device_mac.h b/src/modules/audio_device/main/source/mac/audio_device_mac.h index 6422cd5451..b84f71a35e 100644 --- a/src/modules/audio_device/main/source/mac/audio_device_mac.h +++ b/src/modules/audio_device/main/source/mac/audio_device_mac.h @@ -355,8 +355,6 @@ private: bool _doStopRec; // For rec if not shared device bool _macBookPro; bool _macBookProPanRight; - bool _stereoRender; - bool _stereoRenderRequested; AudioConverterRef _captureConverter; AudioConverterRef _renderConverter; @@ -376,7 +374,6 @@ private: WebRtc_Word32 _renderDelayOffsetSamples; private: - WebRtc_UWord16 _playBufDelay; // playback delay WebRtc_UWord16 _playBufDelayFixed; // fixed playback delay WebRtc_UWord16 _playWarning; diff --git a/src/modules/audio_device/main/source/mac/audio_device_utility_mac.cc b/src/modules/audio_device/main/source/mac/audio_device_utility_mac.cc index d4b349d2ba..f59fd5bb93 100644 --- a/src/modules/audio_device/main/source/mac/audio_device_utility_mac.cc +++ b/src/modules/audio_device/main/source/mac/audio_device_utility_mac.cc @@ -18,8 +18,7 @@ namespace webrtc AudioDeviceUtilityMac::AudioDeviceUtilityMac(const WebRtc_Word32 id) : _critSect(*CriticalSectionWrapper::CreateCriticalSection()), - _id(id), - _lastError(AudioDeviceModule::kAdmErrNone) + _id(id) { WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "%s created", __FUNCTION__); diff --git a/src/modules/audio_device/main/source/mac/audio_device_utility_mac.h b/src/modules/audio_device/main/source/mac/audio_device_utility_mac.h index ccb3d99860..4743e2211b 100644 --- a/src/modules/audio_device/main/source/mac/audio_device_utility_mac.h +++ b/src/modules/audio_device/main/source/mac/audio_device_utility_mac.h @@ -1,5 +1,5 @@ /* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. * * Use of this source code is governed by a BSD-style license * that can be found in the LICENSE file in the root of the source @@ -29,7 +29,6 @@ public: private: CriticalSectionWrapper& _critSect; WebRtc_Word32 _id; - AudioDeviceModule::ErrorCode _lastError; }; } // namespace webrtc diff --git a/src/modules/audio_device/main/test/audio_device_test_api.cc b/src/modules/audio_device/main/test/audio_device_test_api.cc index 238ecb72be..486dbcebb9 100644 --- a/src/modules/audio_device/main/test/audio_device_test_api.cc +++ b/src/modules/audio_device/main/test/audio_device_test_api.cc @@ -76,8 +76,7 @@ class AudioEventObserverAPI: public AudioDeviceObserver { class AudioTransportAPI: public AudioTransport { public: AudioTransportAPI(AudioDeviceModule* audioDevice) - : audio_device_(audioDevice), - rec_count_(0), + : rec_count_(0), play_count_(0) { } @@ -129,7 +128,6 @@ class AudioTransportAPI: public AudioTransport { } private: - AudioDeviceModule* audio_device_; WebRtc_UWord32 rec_count_; WebRtc_UWord32 play_count_; }; diff --git a/src/modules/audio_device/main/test/func_test_manager.cc b/src/modules/audio_device/main/test/func_test_manager.cc index fbf1975081..fddd1d1827 100644 --- a/src/modules/audio_device/main/test/func_test_manager.cc +++ b/src/modules/audio_device/main/test/func_test_manager.cc @@ -68,8 +68,7 @@ const char* GetResource(const char* resource) namespace webrtc { -AudioEventObserver::AudioEventObserver(AudioDeviceModule* audioDevice) : - _audioDevice(audioDevice) +AudioEventObserver::AudioEventObserver(AudioDeviceModule* audioDevice) { } diff --git a/src/modules/audio_device/main/test/func_test_manager.h b/src/modules/audio_device/main/test/func_test_manager.h index ed65bfaed9..e57ac20fea 100644 --- a/src/modules/audio_device/main/test/func_test_manager.h +++ b/src/modules/audio_device/main/test/func_test_manager.h @@ -85,8 +85,6 @@ public: public: ErrorCode _error; WarningCode _warning; -private: - AudioDeviceModule* _audioDevice; }; // ---------------------------------------------------------------------------- diff --git a/src/modules/video_coding/codecs/test_framework/normal_async_test.h b/src/modules/video_coding/codecs/test_framework/normal_async_test.h index 5ce4c8c651..c866217f4c 100644 --- a/src/modules/video_coding/codecs/test_framework/normal_async_test.h +++ b/src/modules/video_coding/codecs/test_framework/normal_async_test.h @@ -38,8 +38,7 @@ class FrameQueue public: FrameQueue() : - _queueRWLock(*webrtc::RWLockWrapper::CreateRWLock()), - _prevTS(-1) + _queueRWLock(*webrtc::RWLockWrapper::CreateRWLock()) { } @@ -56,7 +55,6 @@ public: private: webrtc::RWLockWrapper& _queueRWLock; std::queue _frameBufferQueue; - WebRtc_Word64 _prevTS; }; // feedback signal to encoder diff --git a/src/modules/video_coding/codecs/test_framework/unit_test.h b/src/modules/video_coding/codecs/test_framework/unit_test.h index 71775076b1..0a4fee1468 100644 --- a/src/modules/video_coding/codecs/test_framework/unit_test.h +++ b/src/modules/video_coding/codecs/test_framework/unit_test.h @@ -77,8 +77,6 @@ public: WebRtc_UWord32 decoderSpecificSize = 0, void* decoderSpecificInfo = NULL) : _encodedVideoBuffer(buffer), - _decoderSpecificInfo(decoderSpecificInfo), - _decoderSpecificSize(decoderSpecificSize), _encodeComplete(false) {} WebRtc_Word32 Encoded(webrtc::EncodedImage& encodedImage, const webrtc::CodecSpecificInfo* codecSpecificInfo, @@ -89,8 +87,6 @@ public: webrtc::VideoFrameType EncodedFrameType() const; private: TestVideoEncodedBuffer* _encodedVideoBuffer; - void* _decoderSpecificInfo; - WebRtc_UWord32 _decoderSpecificSize; bool _encodeComplete; webrtc::VideoFrameType _encodedFrameType; }; diff --git a/src/modules/video_coding/main/test/normal_test.h b/src/modules/video_coding/main/test/normal_test.h index 6165d65fa0..982fba4222 100644 --- a/src/modules/video_coding/main/test/normal_test.h +++ b/src/modules/video_coding/main/test/normal_test.h @@ -52,7 +52,6 @@ private: WebRtc_UWord32 _skipCnt; webrtc::VideoCodingModule* _VCMReceiver; webrtc::FrameType _frameType; - WebRtc_UWord8* _payloadData; // max payload size?? WebRtc_UWord16 _seqNo; NormalTest& _test; }; // end of VCMEncodeCompleteCallback diff --git a/src/modules/video_coding/main/test/test_callbacks.h b/src/modules/video_coding/main/test/test_callbacks.h index eae963be91..6731f8ccee 100644 --- a/src/modules/video_coding/main/test/test_callbacks.h +++ b/src/modules/video_coding/main/test/test_callbacks.h @@ -78,7 +78,6 @@ private: float _encodedBytes; VideoCodingModule* _VCMReceiver; FrameType _frameType; - WebRtc_UWord8* _payloadData; WebRtc_UWord16 _seqNo; bool _encodeComplete; WebRtc_Word32 _width; @@ -94,7 +93,6 @@ class VCMRTPEncodeCompleteCallback: public VCMPacketizationCallback public: VCMRTPEncodeCompleteCallback(RtpRtcp* rtp) : _encodedBytes(0), - _seqNo(0), _encodeComplete(false), _RTPModule(rtp) {} @@ -128,8 +126,6 @@ public: private: float _encodedBytes; FrameType _frameType; - WebRtc_UWord8* _payloadData; - WebRtc_UWord16 _seqNo; bool _encodeComplete; RtpRtcp* _RTPModule; WebRtc_Word16 _width; diff --git a/src/modules/video_render/main/source/mac/video_render_nsopengl.h b/src/modules/video_render/main/source/mac/video_render_nsopengl.h index cdd3be74a1..174ae2ad31 100644 --- a/src/modules/video_render/main/source/mac/video_render_nsopengl.h +++ b/src/modules/video_render/main/source/mac/video_render_nsopengl.h @@ -1,5 +1,5 @@ /* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. * * Use of this source code is governed by a BSD-style license * that can be found in the LICENSE file in the root of the source @@ -51,7 +51,7 @@ public: virtual int UpdateSize(int width, int height); - // Setup + // Setup int SetStreamSettings(int streamId, float startWidth, float startHeight, float stopWidth, float stopHeight); int SetStreamCropSettings(int streamId, float startWidth, float startHeight, float stopWidth, float stopHeight); @@ -87,8 +87,6 @@ private: int _stretchedHeight; int _oldStretchedHeight; int _oldStretchedWidth; - int _xOldWidth; - int _yOldHeight; unsigned char* _buffer; int _bufferSize; int _incommingBufferSize; diff --git a/src/modules/video_render/main/source/mac/video_render_nsopengl.mm b/src/modules/video_render/main/source/mac/video_render_nsopengl.mm index 3747cd2ac5..ac95d7055c 100644 --- a/src/modules/video_render/main/source/mac/video_render_nsopengl.mm +++ b/src/modules/video_render/main/source/mac/video_render_nsopengl.mm @@ -34,8 +34,6 @@ _stretchedWidth( 0), _stretchedHeight( 0), _oldStretchedHeight( 0), _oldStretchedWidth( 0), -_xOldWidth( 0), -_yOldHeight( 0), _buffer( 0), _bufferSize( 0), _incommingBufferSize( 0), @@ -426,7 +424,7 @@ int VideoRenderNSOpenGL::ChangeWindow(CocoaRenderView* newWindowRef) return 0; } -/* Check if the thread and event already exist. +/* Check if the thread and event already exist. * If so then they will simply be restarted * If not then create them and continue */ @@ -619,7 +617,7 @@ int VideoRenderNSOpenGL::setRenderTargetFullScreen() [_windowRef setFrame:screenRect]; [_windowRef setBounds:screenRect]; - + _fullScreenWindow = [[CocoaFullScreenWindow alloc]init]; [_fullScreenWindow grabFullScreen]; [[[_fullScreenWindow window] contentView] addSubview:_windowRef]; @@ -655,18 +653,18 @@ VideoRenderNSOpenGL::~VideoRenderNSOpenGL() { if(_fullScreenWindow) { - // Detach CocoaRenderView from full screen view back to + // Detach CocoaRenderView from full screen view back to // it's original parent. [_windowRef removeFromSuperview]; - if(_windowRefSuperView) + if(_windowRefSuperView) { [_windowRefSuperView addSubview:_windowRef]; [_windowRef setFrame:_windowRefSuperViewFrame]; } - + WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, 0, "%s:%d Attempting to release fullscreen window", __FUNCTION__, __LINE__); [_fullScreenWindow releaseFullScreen]; - + } } diff --git a/src/test/testsupport/frame_writer.h b/src/test/testsupport/frame_writer.h index abc5d35191..e91a2992a8 100644 --- a/src/test/testsupport/frame_writer.h +++ b/src/test/testsupport/frame_writer.h @@ -1,5 +1,5 @@ /* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. * * Use of this source code is governed by a BSD-style license * that can be found in the LICENSE file in the root of the source @@ -60,7 +60,6 @@ class FrameWriterImpl : public FrameWriter { private: std::string output_filename_; int frame_length_in_bytes_; - int number_of_frames_; FILE* output_file_; }; diff --git a/src/video_engine/main/test/WindowsTest/windowstest.gypi b/src/video_engine/main/test/WindowsTest/windowstest.gypi index 0bebd98691..83d0515744 100644 --- a/src/video_engine/main/test/WindowsTest/windowstest.gypi +++ b/src/video_engine/main/test/WindowsTest/windowstest.gypi @@ -8,7 +8,9 @@ { 'conditions': [ - ['OS=="win"', { + # TODO(kjellander): Support UseoFMFC on VS2010. + # http://code.google.com/p/webrtc/issues/detail?id=709 + ['OS=="win" and MSVS_VERSION < "2010"', { 'targets': [ # WinTest - GUI test for Windows { @@ -21,10 +23,10 @@ ## VoiceEngine '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine_core', ## VideoEngine - 'video_engine_core', + 'video_engine_core', ], 'include_dirs': [ - './interface', + './interface', '../../../../', # common_types.h and typedefs.h '../commonTestClasses/' ], @@ -34,7 +36,7 @@ 'ChannelDlg.cc', 'ChannelDlg.h', 'ChannelPool.cc', - 'ChannelPool.h', + 'ChannelPool.h', 'renderStartImage.jpg', 'renderTimeoutImage.jpg', 'res\Capture.rc2', @@ -52,7 +54,7 @@ 'CaptureDevicePool.cc', 'tbExternalTransport.h', 'CaptureDevicePool.h', - + ], 'configurations': { 'Common_Base': { diff --git a/src/video_engine/stream_synchronization_unittest.cc b/src/video_engine/stream_synchronization_unittest.cc index d4b002c35e..e0a749400e 100644 --- a/src/video_engine/stream_synchronization_unittest.cc +++ b/src/video_engine/stream_synchronization_unittest.cc @@ -203,7 +203,6 @@ TEST_F(StreamSynchronizationTest, AudioDelay) { int current_audio_delay_ms = 0; int delay_ms = 200; int extra_audio_delay_ms = 0; - int current_extra_delay_ms = 0; int total_video_delay_ms = 0; EXPECT_EQ(0, DelayedVideo(delay_ms, current_audio_delay_ms, @@ -212,7 +211,7 @@ TEST_F(StreamSynchronizationTest, AudioDelay) { // The audio delay is not allowed to change more than this in 1 second. EXPECT_EQ(kMaxAudioDiffMs, extra_audio_delay_ms); current_audio_delay_ms = extra_audio_delay_ms; - current_extra_delay_ms = extra_audio_delay_ms; + int current_extra_delay_ms = extra_audio_delay_ms; send_time_->IncreaseTimeMs(1000); receive_time_->IncreaseTimeMs(800); @@ -273,7 +272,6 @@ TEST_F(StreamSynchronizationTest, BothDelayedVideoLater) { int audio_delay_ms = 100; int video_delay_ms = 300; int extra_audio_delay_ms = 0; - int current_extra_delay_ms = 0; int total_video_delay_ms = 0; EXPECT_EQ(0, DelayedAudioAndVideo(audio_delay_ms, @@ -285,7 +283,7 @@ TEST_F(StreamSynchronizationTest, BothDelayedVideoLater) { // The audio delay is not allowed to change more than this in 1 second. EXPECT_EQ(kMaxAudioDiffMs, extra_audio_delay_ms); current_audio_delay_ms = extra_audio_delay_ms; - current_extra_delay_ms = extra_audio_delay_ms; + int current_extra_delay_ms = extra_audio_delay_ms; send_time_->IncreaseTimeMs(1000); receive_time_->IncreaseTimeMs(800); @@ -358,7 +356,6 @@ TEST_F(StreamSynchronizationTest, BothDelayedAudioLater) { int audio_delay_ms = 300; int video_delay_ms = 100; int extra_audio_delay_ms = 0; - int current_extra_delay_ms = 0; int total_video_delay_ms = 0; EXPECT_EQ(0, DelayedAudioAndVideo(audio_delay_ms, @@ -369,7 +366,6 @@ TEST_F(StreamSynchronizationTest, BothDelayedAudioLater) { EXPECT_EQ(kMaxVideoDiffMs, total_video_delay_ms); EXPECT_EQ(0, extra_audio_delay_ms); current_audio_delay_ms = extra_audio_delay_ms; - current_extra_delay_ms = extra_audio_delay_ms; send_time_->IncreaseTimeMs(1000); receive_time_->IncreaseTimeMs(1000 - std::max(audio_delay_ms, @@ -384,7 +380,6 @@ TEST_F(StreamSynchronizationTest, BothDelayedAudioLater) { EXPECT_EQ(2 * kMaxVideoDiffMs, total_video_delay_ms); EXPECT_EQ(0, extra_audio_delay_ms); current_audio_delay_ms = extra_audio_delay_ms; - current_extra_delay_ms = extra_audio_delay_ms; send_time_->IncreaseTimeMs(1000); receive_time_->IncreaseTimeMs(1000 - std::max(audio_delay_ms, @@ -398,7 +393,6 @@ TEST_F(StreamSynchronizationTest, BothDelayedAudioLater) { &total_video_delay_ms)); EXPECT_EQ(audio_delay_ms - video_delay_ms, total_video_delay_ms); EXPECT_EQ(0, extra_audio_delay_ms); - current_extra_delay_ms = extra_audio_delay_ms; // Simulate that NetEQ introduces some audio delay. current_audio_delay_ms = 50; @@ -415,7 +409,6 @@ TEST_F(StreamSynchronizationTest, BothDelayedAudioLater) { EXPECT_EQ(audio_delay_ms - video_delay_ms + current_audio_delay_ms, total_video_delay_ms); EXPECT_EQ(0, extra_audio_delay_ms); - current_extra_delay_ms = extra_audio_delay_ms; // Simulate that NetEQ reduces its delay. current_audio_delay_ms = 10; diff --git a/src/video_engine/test/auto_test/vie_auto_test.gypi b/src/video_engine/test/auto_test/vie_auto_test.gypi index 8df69e2812..e036441824 100644 --- a/src/video_engine/test/auto_test/vie_auto_test.gypi +++ b/src/video_engine/test/auto_test/vie_auto_test.gypi @@ -99,22 +99,15 @@ 'source/vie_window_manager_factory_win.cc', ], 'conditions': [ - # TODO(andrew): this likely isn't an actual dependency. It should be - # included in webrtc.gyp or video_engine.gyp instead. ['OS=="android"', { 'libraries': [ '-lGLESv2', '-llog', ], }], - ['OS=="win"', { - 'dependencies': [ - 'vie_win_test', - ], - }], ['OS=="linux"', { - # TODO(andrew): these should be provided directly by the projects - # # which require them instead. + # TODO(andrew): These should be provided directly by the projects + # which require them instead. 'libraries': [ '-lXext', '-lX11', diff --git a/src/video_engine/test/libvietest/include/vie_file_capture_device.h b/src/video_engine/test/libvietest/include/vie_file_capture_device.h index 7c986858e0..6348c5a0a4 100644 --- a/src/video_engine/test/libvietest/include/vie_file_capture_device.h +++ b/src/video_engine/test/libvietest/include/vie_file_capture_device.h @@ -50,7 +50,6 @@ class ViEFileCaptureDevice { webrtc::CriticalSectionWrapper* mutex_; WebRtc_UWord32 frame_length_; - WebRtc_UWord8* frame_buffer_; WebRtc_UWord32 width_; WebRtc_UWord32 height_; }; diff --git a/src/voice_engine/test/voice_engine_tests.gypi b/src/voice_engine/test/voice_engine_tests.gypi index de8811b3b6..5e235891c7 100644 --- a/src/voice_engine/test/voice_engine_tests.gypi +++ b/src/voice_engine/test/voice_engine_tests.gypi @@ -104,7 +104,9 @@ }, ], 'conditions': [ - ['OS=="win"', { + # TODO(kjellander): Support UseoFMFC on VS2010. + # http://code.google.com/p/webrtc/issues/detail?id=709 + ['OS=="win" and MSVS_VERSION < "2010"', { 'targets': [ # WinTest - GUI test for Windows {