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Use size_t more consistently for packet/payload lengths.

Browse Source 
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.

This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.

BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom

Review URL: https://webrtc-codereview.appspot.com/23129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pkasting@chromium.org
2014-11-20 22:28:14 +00:00
parent edc6e57a92
commit 4591fbd09f
341 changed files with 2610 additions and 2613 deletions
Download Patch File Download Diff File Expand all files Collapse all files

62
webrtc/modules/audio_coding/main/test/RTPFile.cc
View File

@ -11,6 +11,7 @@
#include "RTPFile.h"
#include <stdlib.h>
#include <limits>
#ifdef WIN32
# include <Winsock2.h>
@ -60,7 +61,7 @@ void RTPStream::MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType,
}
RTPPacket::RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo,
const uint8_t* payloadData, uint16_t payloadSize,
const uint8_t* payloadData, size_t payloadSize,
uint32_t frequency)
: payloadType(payloadType),
timeStamp(timeStamp),
@ -87,7 +88,7 @@ RTPBuffer::~RTPBuffer() {
void 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) {
const size_t payloadSize, uint32_t frequency) {
RTPPacket *packet = new RTPPacket(payloadType, timeStamp, seqNo, payloadData,
payloadSize, frequency);
_queueRWLock->AcquireLockExclusive();
@ -95,8 +96,8 @@ void RTPBuffer::Write(const uint8_t payloadType, const uint32_t timeStamp,
_queueRWLock->ReleaseLockExclusive();
}
uint16_t RTPBuffer::Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
uint16_t payloadSize, uint32_t* offset) {
size_t RTPBuffer::Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
size_t payloadSize, uint32_t* offset) {
_queueRWLock->AcquireLockShared();
RTPPacket *packet = _rtpQueue.front();
_rtpQueue.pop();
@ -143,21 +144,11 @@ void RTPFile::WriteHeader() {
fprintf(_rtpFile, "#!RTPencode%s\n", "1.0");
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;
}
if (fwrite(&dummy_variable, 4, 1, _rtpFile) != 1) {
return;
}
if (fwrite(&dummy_variable, 4, 1, _rtpFile) != 1) {
return;
}
if (fwrite(&dummy_variable, 2, 1, _rtpFile) != 1) {
return;
}
if (fwrite(&dummy_variable, 2, 1, _rtpFile) != 1) {
return;
}
EXPECT_EQ(1u, fwrite(&dummy_variable, 4, 1, _rtpFile));
EXPECT_EQ(1u, fwrite(&dummy_variable, 4, 1, _rtpFile));
EXPECT_EQ(1u, fwrite(&dummy_variable, 4, 1, _rtpFile));
EXPECT_EQ(1u, fwrite(&dummy_variable, 2, 1, _rtpFile));
EXPECT_EQ(1u, fwrite(&dummy_variable, 2, 1, _rtpFile));
fflush(_rtpFile);
}
@ -180,35 +171,26 @@ void RTPFile::ReadHeader() {
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) {
const size_t payloadSize, uint32_t frequency) {
/* write RTP packet to file */
uint8_t rtpHeader[12];
MakeRTPheader(rtpHeader, payloadType, seqNo, timeStamp, 0);
uint16_t lengthBytes = htons(12 + payloadSize + 8);
uint16_t plen = htons(12 + payloadSize);
ASSERT_LE(12 + payloadSize + 8, std::numeric_limits<u_short>::max());
uint16_t lengthBytes = htons(static_cast<u_short>(12 + payloadSize + 8));
uint16_t plen = htons(static_cast<u_short>(12 + payloadSize));
uint32_t offsetMs;
offsetMs = (timeStamp / (frequency / 1000));
offsetMs = htonl(offsetMs);
if (fwrite(&lengthBytes, 2, 1, _rtpFile) != 1) {
return;
}
if (fwrite(&plen, 2, 1, _rtpFile) != 1) {
return;
}
if (fwrite(&offsetMs, 4, 1, _rtpFile) != 1) {
return;
}
if (fwrite(rtpHeader, 12, 1, _rtpFile) != 1) {
return;
}
if (fwrite(payloadData, 1, payloadSize, _rtpFile) != payloadSize) {
return;
}
EXPECT_EQ(1u, fwrite(&lengthBytes, 2, 1, _rtpFile));
EXPECT_EQ(1u, fwrite(&plen, 2, 1, _rtpFile));
EXPECT_EQ(1u, fwrite(&offsetMs, 4, 1, _rtpFile));
EXPECT_EQ(1u, fwrite(&rtpHeader, 12, 1, _rtpFile));
EXPECT_EQ(payloadSize, fwrite(payloadData, 1, payloadSize, _rtpFile));
}
uint16_t RTPFile::Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
uint16_t payloadSize, uint32_t* offset) {
size_t RTPFile::Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
size_t payloadSize, uint32_t* offset) {
uint16_t lengthBytes;
uint16_t plen;
uint8_t rtpHeader[12];
@ -237,7 +219,7 @@ uint16_t RTPFile::Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
if (lengthBytes < 20) {
return 0;
}
if (payloadSize < (lengthBytes - 20)) {
if (payloadSize < static_cast<size_t>((lengthBytes - 20))) {
return 0;
}
lengthBytes -= 20;