Webrtc_Word32 => int32_t in video_coding/main/

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

View File

@ -47,14 +47,14 @@ VCMEncodeCompleteCallback::RegisterTransportCallback(
{
}
WebRtc_Word32
int32_t
VCMEncodeCompleteCallback::SendData(
const FrameType frameType,
const WebRtc_UWord8 payloadType,
const WebRtc_UWord32 timeStamp,
const uint8_t payloadType,
const uint32_t timeStamp,
int64_t capture_time_ms,
const WebRtc_UWord8* payloadData,
const WebRtc_UWord32 payloadSize,
const uint8_t* payloadData,
const uint32_t payloadSize,
const RTPFragmentationHeader& fragmentationHeader,
const RTPVideoHeader* videoHdr)
{
@ -68,8 +68,8 @@ VCMEncodeCompleteCallback::SendData(
rtpInfo.header.markerBit = true; // end of frame
rtpInfo.type.Video.isFirstPacket = true;
rtpInfo.type.Video.codec = _codecType;
rtpInfo.type.Video.height = (WebRtc_UWord16)_height;
rtpInfo.type.Video.width = (WebRtc_UWord16)_width;
rtpInfo.type.Video.height = (uint16_t)_height;
rtpInfo.type.Video.width = (uint16_t)_width;
switch (_codecType)
{
case webrtc::kRTPVideoVP8:
@ -141,14 +141,14 @@ VCMEncodeCompleteCallback::ResetByteCount()
// passes the encoded frame via the RTP module to the decoder
// Packetization callback implementation
WebRtc_Word32
int32_t
VCMRTPEncodeCompleteCallback::SendData(
const FrameType frameType,
const WebRtc_UWord8 payloadType,
const WebRtc_UWord32 timeStamp,
const uint8_t payloadType,
const uint32_t timeStamp,
int64_t capture_time_ms,
const WebRtc_UWord8* payloadData,
const WebRtc_UWord32 payloadSize,
const uint8_t* payloadData,
const uint32_t payloadSize,
const RTPFragmentationHeader& fragmentationHeader,
const RTPVideoHeader* videoHdr)
{
@ -187,7 +187,7 @@ VCMRTPEncodeCompleteCallback::EncodeComplete()
// Decoded Frame Callback Implementation
WebRtc_Word32
int32_t
VCMDecodeCompleteCallback::FrameToRender(I420VideoFrame& videoFrame)
{
if (PrintI420VideoFrame(videoFrame, _decodedFile) < 0) {
@ -198,7 +198,7 @@ VCMDecodeCompleteCallback::FrameToRender(I420VideoFrame& videoFrame)
return VCM_OK;
}
WebRtc_Word32
int32_t
VCMDecodeCompleteCallback::DecodedBytes()
{
return _decodedBytes;
@ -249,7 +249,7 @@ RTPSendCompleteCallback::SendPacket(int channel, const void *data, int len)
if (_rtpDump != NULL)
{
if (_rtpDump->DumpPacket((const WebRtc_UWord8*)data, len) != 0)
if (_rtpDump->DumpPacket((const uint8_t*)data, len) != 0)
{
return -1;
}
@ -268,8 +268,8 @@ RTPSendCompleteCallback::SendPacket(int channel, const void *data, int len)
// Simulate receive time = network delay + packet jitter
// simulated as a Normal distribution random variable with
// mean = networkDelay and variance = jitterVar
WebRtc_Word32
simulatedDelay = (WebRtc_Word32)NormalDist(_networkDelayMs,
int32_t
simulatedDelay = (int32_t)NormalDist(_networkDelayMs,
sqrt(_jitterVar));
newPacket->receiveTime = now + simulatedDelay;
_rtpPackets.push_back(newPacket);
@ -282,7 +282,7 @@ RTPSendCompleteCallback::SendPacket(int channel, const void *data, int len)
{
// Take first packet in list
packet = _rtpPackets.front();
WebRtc_Word64 timeToReceive = packet->receiveTime - now;
int64_t timeToReceive = packet->receiveTime - now;
if (timeToReceive > 0)
{
// No available packets to send
@ -292,7 +292,7 @@ RTPSendCompleteCallback::SendPacket(int channel, const void *data, int len)
_rtpPackets.pop_front();
assert(_rtp); // We must have a configured RTP module for this test.
// Send to receive side
if (_rtp->IncomingPacket((const WebRtc_UWord8*)packet->data,
if (_rtp->IncomingPacket((const uint8_t*)packet->data,
packet->length) < 0)
{
delete packet;
@ -397,16 +397,16 @@ RTPSendCompleteCallback::UnifomLoss(double lossPct)
return randVal < lossPct/100;
}
WebRtc_Word32
PacketRequester::ResendPackets(const WebRtc_UWord16* sequenceNumbers,
WebRtc_UWord16 length)
int32_t
PacketRequester::ResendPackets(const uint16_t* sequenceNumbers,
uint16_t length)
{
return _rtp.SendNACK(sequenceNumbers, length);
}
WebRtc_Word32
SendStatsTest::SendStatistics(const WebRtc_UWord32 bitRate,
const WebRtc_UWord32 frameRate)
int32_t
SendStatsTest::SendStatistics(const uint32_t bitRate,
const uint32_t frameRate)
{
TEST(frameRate <= _framerate);
TEST(bitRate > _bitrate / 2 && bitRate < 3 * _bitrate / 2);
@ -414,7 +414,7 @@ SendStatsTest::SendStatistics(const WebRtc_UWord32 bitRate,
return 0;
}
WebRtc_Word32 KeyFrameReqTest::RequestKeyFrame() {
int32_t KeyFrameReqTest::RequestKeyFrame() {
printf("Key frame requested\n");
return 0;
}
@ -433,13 +433,13 @@ VideoProtectionCallback::~VideoProtectionCallback()
//
}
WebRtc_Word32
int32_t
VideoProtectionCallback::ProtectionRequest(
const FecProtectionParams* delta_fec_params,
const FecProtectionParams* key_fec_params,
WebRtc_UWord32* sent_video_rate_bps,
WebRtc_UWord32* sent_nack_rate_bps,
WebRtc_UWord32* sent_fec_rate_bps)
uint32_t* sent_video_rate_bps,
uint32_t* sent_nack_rate_bps,
uint32_t* sent_fec_rate_bps)
{
key_fec_params_ = *key_fec_params;
delta_fec_params_ = *delta_fec_params;