Use int64_t more consistently for times, in particular for RTT values.

Existing code was inconsistent about whether to use uint16_t, int, unsigned int,
or uint32_t, and sometimes silently truncated one to another, or truncated
int64_t.  Because most core time-handling functions use int64_t, being
consistent about using int64_t unless otherwise necessary minimizes the number
of explicit or implicit casts.

BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, holmer@google.com, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pkasting@chromium.org
2015-01-12 21:51:21 +00:00
parent a7add19cf4
commit 16825b1a82
124 changed files with 422 additions and 417 deletions

View File

@ -181,10 +181,10 @@ int32_t ModuleRtpRtcpImpl::Process() {
last_rtt_process_time_ && process_rtt) {
std::vector<RTCPReportBlock> receive_blocks;
rtcp_receiver_.StatisticsReceived(&receive_blocks);
uint16_t max_rtt = 0;
int64_t max_rtt = 0;
for (std::vector<RTCPReportBlock>::iterator it = receive_blocks.begin();
it != receive_blocks.end(); ++it) {
uint16_t rtt = 0;
int64_t rtt = 0;
rtcp_receiver_.RTT(it->remoteSSRC, &rtt, NULL, NULL, NULL);
max_rtt = (rtt > max_rtt) ? rtt : max_rtt;
}
@ -216,7 +216,7 @@ int32_t ModuleRtpRtcpImpl::Process() {
} else {
// Report rtt from receiver.
if (process_rtt) {
uint16_t rtt_ms;
int64_t rtt_ms;
if (rtt_stats_ && rtcp_receiver_.GetAndResetXrRrRtt(&rtt_ms)) {
rtt_stats_->OnRttUpdate(rtt_ms);
}
@ -707,7 +707,7 @@ void ModuleRtpRtcpImpl::SetRTCPStatus(const RTCPMethod method) {
// Only for internal test.
uint32_t ModuleRtpRtcpImpl::LastSendReport(
uint32_t& last_rtcptime) {
int64_t& last_rtcptime) {
return rtcp_sender_.LastSendReport(last_rtcptime);
}
@ -747,14 +747,14 @@ int32_t ModuleRtpRtcpImpl::RemoteNTP(
// Get RoundTripTime.
int32_t ModuleRtpRtcpImpl::RTT(const uint32_t remote_ssrc,
uint16_t* rtt,
uint16_t* avg_rtt,
uint16_t* min_rtt,
uint16_t* max_rtt) const {
int64_t* rtt,
int64_t* avg_rtt,
int64_t* min_rtt,
int64_t* max_rtt) const {
int32_t ret = rtcp_receiver_.RTT(remote_ssrc, rtt, avg_rtt, min_rtt, max_rtt);
if (rtt && *rtt == 0) {
// Try to get RTT from RtcpRttStats class.
*rtt = static_cast<uint16_t>(rtt_ms());
*rtt = rtt_ms();
}
return ret;
}
@ -944,7 +944,7 @@ int32_t ModuleRtpRtcpImpl::SendNACK(const uint16_t* nack_list,
bool ModuleRtpRtcpImpl::TimeToSendFullNackList(int64_t now) const {
// Use RTT from RtcpRttStats class if provided.
uint16_t rtt = rtt_ms();
int64_t rtt = rtt_ms();
if (rtt == 0) {
rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), NULL, &rtt, NULL, NULL);
}
@ -1244,7 +1244,7 @@ int32_t ModuleRtpRtcpImpl::SendRTCPReferencePictureSelection(
GetFeedbackState(), kRtcpRpsi, 0, 0, false, picture_id);
}
uint32_t ModuleRtpRtcpImpl::SendTimeOfSendReport(
int64_t ModuleRtpRtcpImpl::SendTimeOfSendReport(
const uint32_t send_report) {
return rtcp_sender_.SendTimeOfSendReport(send_report);
}
@ -1261,7 +1261,7 @@ void ModuleRtpRtcpImpl::OnReceivedNACK(
return;
}
// Use RTT from RtcpRttStats class if provided.
uint16_t rtt = rtt_ms();
int64_t rtt = rtt_ms();
if (rtt == 0) {
rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), NULL, &rtt, NULL, NULL);
}
@ -1324,12 +1324,12 @@ void ModuleRtpRtcpImpl::SetRtcpReceiverSsrcs(uint32_t main_ssrc) {
rtcp_receiver_.SetSsrcs(main_ssrc, ssrcs);
}
void ModuleRtpRtcpImpl::set_rtt_ms(uint32_t rtt_ms) {
void ModuleRtpRtcpImpl::set_rtt_ms(int64_t rtt_ms) {
CriticalSectionScoped cs(critical_section_rtt_.get());
rtt_ms_ = rtt_ms;
}
uint32_t ModuleRtpRtcpImpl::rtt_ms() const {
int64_t ModuleRtpRtcpImpl::rtt_ms() const {
CriticalSectionScoped cs(critical_section_rtt_.get());
return rtt_ms_;
}