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Use size_t more consistently for packet/payload lengths.

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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

8
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -259,7 +259,7 @@ void ModuleRtpRtcpImpl::SetRtxSendPayloadType(int payload_type) {
int32_t ModuleRtpRtcpImpl::IncomingRtcpPacket(
const uint8_t* rtcp_packet,
const uint16_t length) {
const size_t length) {
// Allow receive of non-compound RTCP packets.
RTCPUtility::RTCPParserV2 rtcp_parser(rtcp_packet, length, true);
@ -504,7 +504,7 @@ int32_t ModuleRtpRtcpImpl::SendOutgoingData(
uint32_t time_stamp,
int64_t capture_time_ms,
const uint8_t* payload_data,
uint32_t payload_size,
size_t payload_size,
const RTPFragmentationHeader* fragmentation,
const RTPVideoHeader* rtp_video_hdr) {
rtcp_sender_.SetLastRtpTime(time_stamp, capture_time_ms);
@ -603,7 +603,7 @@ bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc,
return true;
}
int ModuleRtpRtcpImpl::TimeToSendPadding(int bytes) {
size_t ModuleRtpRtcpImpl::TimeToSendPadding(size_t bytes) {
if (!IsDefaultModule()) {
// Don't send from default module.
return rtp_sender_.TimeToSendPadding(bytes);
@ -816,7 +816,7 @@ bool ModuleRtpRtcpImpl::RtcpXrRrtrStatus() const {
}
int32_t ModuleRtpRtcpImpl::DataCountersRTP(
uint32_t* bytes_sent,
size_t* bytes_sent,
uint32_t* packets_sent) const {
StreamDataCounters rtp_stats;
StreamDataCounters rtx_stats;