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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

14
webrtc/modules/utility/source/rtp_dump_impl.cc
View File

@ -12,6 +12,7 @@
#include <assert.h>
#include <stdio.h>
#include <limits>
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/logging.h"
@ -145,7 +146,7 @@ bool RtpDumpImpl::IsActive() const
return _file.Open();
}
int32_t RtpDumpImpl::DumpPacket(const uint8_t* packet, uint16_t packetLength)
int32_t RtpDumpImpl::DumpPacket(const uint8_t* packet, size_t packetLength)
{
CriticalSectionScoped lock(_critSect);
if (!IsActive())
@ -158,7 +159,9 @@ int32_t RtpDumpImpl::DumpPacket(const uint8_t* packet, uint16_t packetLength)
return -1;
}
if (packetLength < 1)
rtpDumpPktHdr_t hdr;
size_t total_size = packetLength + sizeof hdr;
if (packetLength < 1 || total_size > std::numeric_limits<uint16_t>::max())
{
return -1;
}
@ -167,11 +170,8 @@ int32_t RtpDumpImpl::DumpPacket(const uint8_t* packet, uint16_t packetLength)
// considered RTP (without further verification).
bool isRTCP = RTCP(packet);
rtpDumpPktHdr_t hdr;
uint32_t offset;
// Offset is relative to when recording was started.
offset = GetTimeInMS();
uint32_t offset = GetTimeInMS();
if (offset < _startTime)
{
// Compensate for wraparound.
@ -181,7 +181,7 @@ int32_t RtpDumpImpl::DumpPacket(const uint8_t* packet, uint16_t packetLength)
}
hdr.offset = RtpDumpHtonl(offset);
hdr.length = RtpDumpHtons((uint16_t)(packetLength + sizeof(hdr)));
hdr.length = RtpDumpHtons((uint16_t)(total_size));
if (isRTCP)
{
hdr.plen = 0;