Use size_t more consistently for packet/payload lengths.
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
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 6446cb7..db10956 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -259,7 +259,7 @@
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 @@
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 @@
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 @@
}
int32_t ModuleRtpRtcpImpl::DataCountersRTP(
- uint32_t* bytes_sent,
+ size_t* bytes_sent,
uint32_t* packets_sent) const {
StreamDataCounters rtp_stats;
StreamDataCounters rtx_stats;
