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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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78
webrtc/modules/audio_coding/main/test/RTPFile.cc
View File

@ -25,23 +25,23 @@
namespace webrtc {
void RTPStream::ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const WebRtc_UWord8* rtpHeader)
void RTPStream::ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const uint8_t* rtpHeader)
{
rtpInfo->header.payloadType = rtpHeader[1];
rtpInfo->header.sequenceNumber = (static_cast<WebRtc_UWord16>(rtpHeader[2])<<8) | rtpHeader[3];
rtpInfo->header.timestamp = (static_cast<WebRtc_UWord32>(rtpHeader[4])<<24) |
(static_cast<WebRtc_UWord32>(rtpHeader[5])<<16) |
(static_cast<WebRtc_UWord32>(rtpHeader[6])<<8) |
rtpInfo->header.sequenceNumber = (static_cast<uint16_t>(rtpHeader[2])<<8) | rtpHeader[3];
rtpInfo->header.timestamp = (static_cast<uint32_t>(rtpHeader[4])<<24) |
(static_cast<uint32_t>(rtpHeader[5])<<16) |
(static_cast<uint32_t>(rtpHeader[6])<<8) |
rtpHeader[7];
rtpInfo->header.ssrc = (static_cast<WebRtc_UWord32>(rtpHeader[8])<<24) |
(static_cast<WebRtc_UWord32>(rtpHeader[9])<<16) |
(static_cast<WebRtc_UWord32>(rtpHeader[10])<<8) |
rtpInfo->header.ssrc = (static_cast<uint32_t>(rtpHeader[8])<<24) |
(static_cast<uint32_t>(rtpHeader[9])<<16) |
(static_cast<uint32_t>(rtpHeader[10])<<8) |
rtpHeader[11];
}
void RTPStream::MakeRTPheader(WebRtc_UWord8* rtpHeader,
WebRtc_UWord8 payloadType, WebRtc_Word16 seqNo,
WebRtc_UWord32 timeStamp, WebRtc_UWord32 ssrc)
void RTPStream::MakeRTPheader(uint8_t* rtpHeader,
uint8_t payloadType, int16_t seqNo,
uint32_t timeStamp, uint32_t ssrc)
{
rtpHeader[0]=(unsigned char)0x80;
rtpHeader[1]=(unsigned char)(payloadType & 0xFF);
@ -61,9 +61,9 @@ void RTPStream::MakeRTPheader(WebRtc_UWord8* rtpHeader,
}
RTPPacket::RTPPacket(WebRtc_UWord8 payloadType, WebRtc_UWord32 timeStamp,
WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency)
RTPPacket::RTPPacket(uint8_t payloadType, uint32_t timeStamp,
int16_t seqNo, const uint8_t* payloadData,
uint16_t payloadSize, uint32_t frequency)
:
payloadType(payloadType),
timeStamp(timeStamp),
@ -73,7 +73,7 @@ frequency(frequency)
{
if (payloadSize > 0)
{
this->payloadData = new WebRtc_UWord8[payloadSize];
this->payloadData = new uint8_t[payloadSize];
memcpy(this->payloadData, payloadData, payloadSize);
}
}
@ -94,9 +94,9 @@ RTPBuffer::~RTPBuffer()
}
void
RTPBuffer::Write(const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
const WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
const WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency)
RTPBuffer::Write(const uint8_t payloadType, const uint32_t timeStamp,
const int16_t seqNo, const uint8_t* payloadData,
const uint16_t payloadSize, uint32_t frequency)
{
RTPPacket *packet = new RTPPacket(payloadType, timeStamp, seqNo, payloadData, payloadSize, frequency);
_queueRWLock->AcquireLockExclusive();
@ -104,11 +104,11 @@ RTPBuffer::Write(const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp
_queueRWLock->ReleaseLockExclusive();
}
WebRtc_UWord16
uint16_t
RTPBuffer::Read(WebRtcRTPHeader* rtpInfo,
WebRtc_UWord8* payloadData,
WebRtc_UWord16 payloadSize,
WebRtc_UWord32* offset)
uint8_t* payloadData,
uint16_t payloadSize,
uint32_t* offset)
{
_queueRWLock->AcquireLockShared();
RTPPacket *packet = _rtpQueue.front();
@ -165,7 +165,7 @@ void RTPFile::WriteHeader()
{
// Write data in a format that NetEQ and RTP Play can parse
fprintf(_rtpFile, "#!RTPencode%s\n", "1.0");
WebRtc_UWord32 dummy_variable = 0;
uint32_t dummy_variable = 0;
// should be converted to network endian format, but does not matter when 0
if (fwrite(&dummy_variable, 4, 1, _rtpFile) != 1) {
return;
@ -187,8 +187,8 @@ void RTPFile::WriteHeader()
void RTPFile::ReadHeader()
{
WebRtc_UWord32 start_sec, start_usec, source;
WebRtc_UWord16 port, padding;
uint32_t start_sec, start_usec, source;
uint16_t port, padding;
char fileHeader[40];
EXPECT_TRUE(fgets(fileHeader, 40, _rtpFile) != 0);
EXPECT_EQ(1u, fread(&start_sec, 4, 1, _rtpFile));
@ -203,16 +203,16 @@ void RTPFile::ReadHeader()
padding=ntohs(padding);
}
void RTPFile::Write(const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
const WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
const WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency)
void RTPFile::Write(const uint8_t payloadType, const uint32_t timeStamp,
const int16_t seqNo, const uint8_t* payloadData,
const uint16_t payloadSize, uint32_t frequency)
{
/* write RTP packet to file */
WebRtc_UWord8 rtpHeader[12];
uint8_t rtpHeader[12];
MakeRTPheader(rtpHeader, payloadType, seqNo, timeStamp, 0);
WebRtc_UWord16 lengthBytes = htons(12 + payloadSize + 8);
WebRtc_UWord16 plen = htons(12 + payloadSize);
WebRtc_UWord32 offsetMs;
uint16_t lengthBytes = htons(12 + payloadSize + 8);
uint16_t plen = htons(12 + payloadSize);
uint32_t offsetMs;
offsetMs = (timeStamp/(frequency/1000));
offsetMs = htonl(offsetMs);
@ -233,14 +233,14 @@ void RTPFile::Write(const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeSt
}
}
WebRtc_UWord16 RTPFile::Read(WebRtcRTPHeader* rtpInfo,
WebRtc_UWord8* payloadData,
WebRtc_UWord16 payloadSize,
WebRtc_UWord32* offset)
uint16_t RTPFile::Read(WebRtcRTPHeader* rtpInfo,
uint8_t* payloadData,
uint16_t payloadSize,
uint32_t* offset)
{
WebRtc_UWord16 lengthBytes;
WebRtc_UWord16 plen;
WebRtc_UWord8 rtpHeader[12];
uint16_t lengthBytes;
uint16_t plen;
uint8_t rtpHeader[12];
size_t read_len = fread(&lengthBytes, 2, 1, _rtpFile);
/* Check if we have reached end of file. */
if ((read_len == 0) && feof(_rtpFile))