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Prepare to roll Chromium to 149181.

Browse Source 
- This roll brings in VS2010 by default. The buildbots
  need updating (issue710).
- We'll roll to 149181 later (past current Canary) to fix
  a Mac gyp issue:
  https://chromiumcodereview.appspot.com/10824105
- Chromium is now using a later libvpx than us. We should
  investigate rolling our standalone build.
- Fix set-but-unused-warning
- Fix -Wunused-private-field warnings on Mac.

TBR=kjellander@webrtc.org
BUG=issue709,issue710
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/709007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2544 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andrew@webrtc.org
2012-08-01 01:40:02 +00:00
parent bf853918eb
commit d7a71d0719
26 changed files with 145 additions and 199 deletions
Download Patch File Download Diff File Expand all files Collapse all files

122
src/modules/audio_coding/main/test/APITest.cc
View File

@@ -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<APITest*>(obj)->PushAudioRunA();
}
bool
bool
APITest::PushAudioThreadB(void* obj)
{
return static_cast<APITest*>(obj)->PushAudioRunB();
}
bool
bool
APITest::PullAudioThreadA(void* obj)
{
return static_cast<APITest*>(obj)->PullAudioRunA();
}
bool
bool
APITest::PullAudioThreadB(void* obj)
{
return static_cast<APITest*>(obj)->PullAudioRunB();
@@ -409,7 +407,7 @@ APITest::APIThreadB(void* obj)
return static_cast<APITest*>(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);
}

34
src/modules/audio_coding/main/test/APITest.h
View File

@@ -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;

59
src/modules/audio_coding/main/test/Channel.cc
View File

@@ -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

23
src/modules/audio_coding/main/test/Channel.h
View File

@@ -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;
};

2
src/modules/audio_coding/main/test/EncodeDecodeTest.cc
View File

@@ -50,8 +50,6 @@ Sender::Sender()
: _acm(NULL),
_pcmFile(),
_audioFrame(),
_payloadSize(0),
_timeStamp(0),
_packetization(NULL) {
}

5
src/modules/audio_coding/main/test/EncodeDecodeTest.h
View File

@@ -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

1
src/modules/audio_coding/main/test/TestStereo.h
View File

@@ -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_;

5
src/modules/audio_coding/main/test/TwoWayCommunication.h
View File

@@ -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;
};

3
src/modules/audio_device/main/source/mac/audio_device_mac.h
View File

@@ -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;

3
src/modules/audio_device/main/source/mac/audio_device_utility_mac.cc
View File

@@ -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__);

3
src/modules/audio_device/main/source/mac/audio_device_utility_mac.h
View File

@@ -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

4
src/modules/audio_device/main/test/audio_device_test_api.cc
View File

@@ -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_;
};

3
src/modules/audio_device/main/test/func_test_manager.cc
View File

@@ -68,8 +68,7 @@ const char* GetResource(const char* resource)
namespace webrtc
{
AudioEventObserver::AudioEventObserver(AudioDeviceModule* audioDevice) :
_audioDevice(audioDevice)
AudioEventObserver::AudioEventObserver(AudioDeviceModule* audioDevice)
{
}

2
src/modules/audio_device/main/test/func_test_manager.h
View File

@@ -85,8 +85,6 @@ public:
public:
ErrorCode _error;
WarningCode _warning;
private:
AudioDeviceModule* _audioDevice;
};
// ----------------------------------------------------------------------------

4
src/modules/video_coding/codecs/test_framework/normal_async_test.h
View File

@@ -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<FrameQueueTuple *> _frameBufferQueue;
WebRtc_Word64 _prevTS;
};
// feedback signal to encoder

4
src/modules/video_coding/codecs/test_framework/unit_test.h
View File

@@ -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;
};

1
src/modules/video_coding/main/test/normal_test.h
View File

@@ -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

4
src/modules/video_coding/main/test/test_callbacks.h
View File

@@ -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;

6
src/modules/video_render/main/source/mac/video_render_nsopengl.h
View File

@@ -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;

14
src/modules/video_render/main/source/mac/video_render_nsopengl.mm
View File

@@ -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];
}
}

3
src/test/testsupport/frame_writer.h
View File

@@ -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_;
};

12
src/video_engine/main/test/WindowsTest/windowstest.gypi
View File

@@ -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': {

11
src/video_engine/stream_synchronization_unittest.cc
View File

@@ -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;

11
src/video_engine/test/auto_test/vie_auto_test.gypi
View File

@@ -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',

1
src/video_engine/test/libvietest/include/vie_file_capture_device.h
View File

@@ -50,7 +50,6 @@ class ViEFileCaptureDevice {
webrtc::CriticalSectionWrapper* mutex_;
WebRtc_UWord32 frame_length_;
WebRtc_UWord8* frame_buffer_;
WebRtc_UWord32 width_;
WebRtc_UWord32 height_;
};

4
src/voice_engine/test/voice_engine_tests.gypi
View File

@@ -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
{