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Webrtc_Word32 => int32_t in video_coding/main/

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BUG=

Review URL: https://webrtc-codereview.appspot.com/1279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3753 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org
2013-04-02 15:54:38 +00:00
parent cfc07c943f
commit 7b859cc1e9
67 changed files with 1130 additions and 1130 deletions
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48
webrtc/modules/video_coding/main/test/rtp_player.cc
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@ -178,7 +178,7 @@ RTPPlayer::~RTPPlayer()
}
}
WebRtc_Word32 RTPPlayer::Initialize(const PayloadTypeList* payloadList)
int32_t RTPPlayer::Initialize(const PayloadTypeList* payloadList)
{
RtpRtcp::Configuration configuration;
configuration.id = 1;
@ -192,7 +192,7 @@ WebRtc_Word32 RTPPlayer::Initialize(const PayloadTypeList* payloadList)
_randVec[i] = rand();
}
_randVecPos = 0;
WebRtc_Word32 ret = _rtpModule->SetNACKStatus(kNackOff,
int32_t ret = _rtpModule->SetNACKStatus(kNackOff,
kMaxPacketAgeToNack);
if (ret < 0)
{
@ -229,7 +229,7 @@ WebRtc_Word32 RTPPlayer::Initialize(const PayloadTypeList* payloadList)
return 0;
}
WebRtc_Word32 RTPPlayer::ReadHeader()
int32_t RTPPlayer::ReadHeader()
{
char firstline[FIRSTLINELEN];
if (_rtpFile == NULL)
@ -254,11 +254,11 @@ WebRtc_Word32 RTPPlayer::ReadHeader()
return -1;
}
WebRtc_UWord32 start_sec;
WebRtc_UWord32 start_usec;
WebRtc_UWord32 source;
WebRtc_UWord16 port;
WebRtc_UWord16 padding;
uint32_t start_sec;
uint32_t start_usec;
uint32_t source;
uint16_t port;
uint16_t padding;
EXPECT_GT(fread(&start_sec, 4, 1, _rtpFile), 0u);
start_sec=ntohl(start_sec);
@ -273,18 +273,18 @@ WebRtc_Word32 RTPPlayer::ReadHeader()
return 0;
}
WebRtc_UWord32 RTPPlayer::TimeUntilNextPacket() const
uint32_t RTPPlayer::TimeUntilNextPacket() const
{
WebRtc_Word64 timeLeft = (_nextRtpTime - _firstPacketRtpTime) -
int64_t timeLeft = (_nextRtpTime - _firstPacketRtpTime) -
(_clock->TimeInMilliseconds() - _firstPacketTimeMs);
if (timeLeft < 0)
{
return 0;
}
return static_cast<WebRtc_UWord32>(timeLeft);
return static_cast<uint32_t>(timeLeft);
}
WebRtc_Word32 RTPPlayer::NextPacket(const WebRtc_Word64 timeNow)
int32_t RTPPlayer::NextPacket(const int64_t timeNow)
{
// Send any packets ready to be resent,
RawRtpPacket* resend_packet = _lostPackets.NextPacketToResend(timeNow);
@ -309,15 +309,15 @@ WebRtc_Word32 RTPPlayer::NextPacket(const WebRtc_Word64 timeNow)
_rtpModule->Process();
if (_firstPacket)
{
_firstPacketRtpTime = static_cast<WebRtc_Word64>(_nextRtpTime);
_firstPacketRtpTime = static_cast<int64_t>(_nextRtpTime);
_firstPacketTimeMs = _clock->TimeInMilliseconds();
}
if (_reordering && _reorderBuffer == NULL)
{
_reorderBuffer = new RawRtpPacket(reinterpret_cast<WebRtc_UWord8*>(_nextPacket), static_cast<WebRtc_UWord16>(_nextPacketLength));
_reorderBuffer = new RawRtpPacket(reinterpret_cast<uint8_t*>(_nextPacket), static_cast<uint16_t>(_nextPacketLength));
return 0;
}
WebRtc_Word32 ret = SendPacket(reinterpret_cast<WebRtc_UWord8*>(_nextPacket), static_cast<WebRtc_UWord16>(_nextPacketLength));
int32_t ret = SendPacket(reinterpret_cast<uint8_t*>(_nextPacket), static_cast<uint16_t>(_nextPacketLength));
if (_reordering && _reorderBuffer != NULL)
{
RawRtpPacket* rtpPacket = _reorderBuffer;
@ -348,14 +348,14 @@ WebRtc_Word32 RTPPlayer::NextPacket(const WebRtc_Word64 timeNow)
return 0;
}
WebRtc_Word32 RTPPlayer::SendPacket(WebRtc_UWord8* rtpData, WebRtc_UWord16 rtpLen)
int32_t RTPPlayer::SendPacket(uint8_t* rtpData, uint16_t rtpLen)
{
if ((_randVec[(_randVecPos++) % RAND_VEC_LENGTH] + 1.0)/(RAND_MAX + 1.0) < _lossRate &&
_noLossStartup < 0)
{
if (_nackEnabled)
{
const WebRtc_UWord16 seqNo = (rtpData[2] << 8) + rtpData[3];
const uint16_t seqNo = (rtpData[2] << 8) + rtpData[3];
printf("Throw: %u\n", seqNo);
_lostPackets.AddPacket(new RawRtpPacket(rtpData, rtpLen));
return 0;
@ -363,7 +363,7 @@ WebRtc_Word32 RTPPlayer::SendPacket(WebRtc_UWord8* rtpData, WebRtc_UWord16 rtpLe
}
else if (rtpLen > 0)
{
WebRtc_Word32 ret = _rtpModule->IncomingPacket(rtpData, rtpLen);
int32_t ret = _rtpModule->IncomingPacket(rtpData, rtpLen);
if (ret < 0)
{
return -1;
@ -376,9 +376,9 @@ WebRtc_Word32 RTPPlayer::SendPacket(WebRtc_UWord8* rtpData, WebRtc_UWord16 rtpLe
return 1;
}
WebRtc_Word32 RTPPlayer::ReadPacket(WebRtc_Word16* rtpdata, WebRtc_UWord32* offset)
int32_t RTPPlayer::ReadPacket(int16_t* rtpdata, uint32_t* offset)
{
WebRtc_UWord16 length, plen;
uint16_t length, plen;
if (fread(&length,2,1,_rtpFile)==0)
return(-1);
@ -393,7 +393,7 @@ WebRtc_Word32 RTPPlayer::ReadPacket(WebRtc_Word16* rtpdata, WebRtc_UWord32* offs
*offset=ntohl(*offset);
// Use length here because a plen of 0 specifies rtcp
length = (WebRtc_UWord16) (length - HDR_SIZE);
length = (uint16_t) (length - HDR_SIZE);
if (fread((unsigned short *) rtpdata,1,length,_rtpFile) != length)
return(-1);
@ -408,7 +408,7 @@ WebRtc_Word32 RTPPlayer::ReadPacket(WebRtc_Word16* rtpdata, WebRtc_UWord32* offs
return plen;
}
WebRtc_Word32 RTPPlayer::SimulatePacketLoss(float lossRate, bool enableNack, WebRtc_UWord32 rttMs)
int32_t RTPPlayer::SimulatePacketLoss(float lossRate, bool enableNack, uint32_t rttMs)
{
_nackEnabled = enableNack;
_lossRate = lossRate;
@ -416,13 +416,13 @@ WebRtc_Word32 RTPPlayer::SimulatePacketLoss(float lossRate, bool enableNack, Web
return 0;
}
WebRtc_Word32 RTPPlayer::SetReordering(bool enabled)
int32_t RTPPlayer::SetReordering(bool enabled)
{
_reordering = enabled;
return 0;
}
WebRtc_Word32 RTPPlayer::ResendPackets(const WebRtc_UWord16* sequenceNumbers, WebRtc_UWord16 length)
int32_t RTPPlayer::ResendPackets(const uint16_t* sequenceNumbers, uint16_t length)
{
if (sequenceNumbers == NULL)
{