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WebRtc_Word32 => int32_t etc. in audio_coding/

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BUG=314

Review URL: https://webrtc-codereview.appspot.com/1271006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3789 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org
2013-04-09 00:28:06 +00:00
parent 6faf71d27b
commit 0946a56023
382 changed files with 8469 additions and 8488 deletions
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42
webrtc/modules/audio_coding/main/test/APITest.cc
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@ -50,7 +50,7 @@ namespace webrtc {
void
APITest::Wait(WebRtc_UWord32 waitLengthMs)
APITest::Wait(uint32_t waitLengthMs)
{
if(_randomTest)
{
@ -160,19 +160,19 @@ APITest::~APITest()
//WebRtc_Word16
//APITest::SetInFile(char* fileName, WebRtc_UWord16 frequencyHz)
//int16_t
//APITest::SetInFile(char* fileName, uint16_t frequencyHz)
//{
// return _inFile.Open(fileName, frequencyHz, "rb");
//}
//
//WebRtc_Word16
//APITest::SetOutFile(char* fileName, WebRtc_UWord16 frequencyHz)
//int16_t
//APITest::SetOutFile(char* fileName, uint16_t frequencyHz)
//{
// return _outFile.Open(fileName, frequencyHz, "wb");
//}
WebRtc_Word16
int16_t
APITest::SetUp()
{
_acmA = AudioCodingModule::Create(1);
@ -181,8 +181,8 @@ APITest::SetUp()
CodecInst dummyCodec;
int lastPayloadType = 0;
WebRtc_Word16 numCodecs = _acmA->NumberOfCodecs();
for(WebRtc_UWord8 n = 0; n < numCodecs; n++)
int16_t numCodecs = _acmA->NumberOfCodecs();
for(uint8_t n = 0; n < numCodecs; n++)
{
AudioCodingModule::Codec(n, &dummyCodec);
if((STR_CASE_CMP(dummyCodec.plname, "CN") == 0) &&
@ -250,15 +250,15 @@ APITest::SetUp()
_thereIsDecoderB = true;
// Register Send Codec
AudioCodingModule::Codec((WebRtc_UWord8)_codecCntrA, &dummyCodec);
AudioCodingModule::Codec((uint8_t)_codecCntrA, &dummyCodec);
CHECK_ERROR_MT(_acmA->RegisterSendCodec(dummyCodec));
_thereIsEncoderA = true;
//
AudioCodingModule::Codec((WebRtc_UWord8)_codecCntrB, &dummyCodec);
AudioCodingModule::Codec((uint8_t)_codecCntrB, &dummyCodec);
CHECK_ERROR_MT(_acmB->RegisterSendCodec(dummyCodec));
_thereIsEncoderB = true;
WebRtc_UWord16 frequencyHz;
uint16_t frequencyHz;
printf("\n\nAPI Test\n");
printf("========\n");
@ -747,8 +747,8 @@ APITest::Perform()
// Keep main thread waiting for sender/receiver
// threads to complete
EventWrapper* completeEvent = EventWrapper::Create();
WebRtc_UWord64 startTime = TickTime::MillisecondTimestamp();
WebRtc_UWord64 currentTime;
uint64_t startTime = TickTime::MillisecondTimestamp();
uint64_t currentTime;
do
{
{
@ -891,11 +891,11 @@ APITest::TestDelay(char side)
{
AudioCodingModule* myACM;
Channel* myChannel;
WebRtc_Word32* myMinDelay;
int32_t* myMinDelay;
EventWrapper* myEvent = EventWrapper::Create();
WebRtc_UWord32 inTimestamp = 0;
WebRtc_UWord32 outTimestamp = 0;
uint32_t inTimestamp = 0;
uint32_t outTimestamp = 0;
double estimDelay = 0;
double averageEstimDelay = 0;
@ -937,7 +937,7 @@ APITest::TestDelay(char side)
CHECK_ERROR_MT(myACM->PlayoutTimestamp(&outTimestamp));
//std::cout << outTimestamp << std::endl << std::flush;
estimDelay = (double)((WebRtc_UWord32)(inTimestamp - outTimestamp)) /
estimDelay = (double)((uint32_t)(inTimestamp - outTimestamp)) /
((double)myACM->ReceiveFrequency() / 1000.0);
estimDelayCB.Update(estimDelay);
@ -1063,7 +1063,7 @@ APITest::TestRegisteration(char sendSide)
if(!FixedPayloadTypeCodec(myCodec.plname))
{
WebRtc_Word32 i;
int32_t i;
for(i = 0; i < 32; i++)
{
if(!_payloadUsed[i])
@ -1172,8 +1172,8 @@ APITest::TestPlayout(char receiveSide)
receiveACM = _acmA;
}
WebRtc_Word32 receiveFreqHz = receiveACM->ReceiveFrequency();
WebRtc_Word32 playoutFreqHz = receiveACM->PlayoutFrequency();
int32_t receiveFreqHz = receiveACM->ReceiveFrequency();
int32_t playoutFreqHz = receiveACM->PlayoutFrequency();
CHECK_ERROR_MT(receiveFreqHz);
CHECK_ERROR_MT(playoutFreqHz);
@ -1437,7 +1437,7 @@ APITest::ChangeCodec(char side)
{
CodecInst myCodec;
AudioCodingModule* myACM;
WebRtc_UWord8* codecCntr;
uint8_t* codecCntr;
bool* thereIsEncoder;
bool* vad;
bool* dtx;