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Add a default RTT to CallStats and use different values for buffered/real-time mode.

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BUG=1613

Review URL: https://webrtc-codereview.appspot.com/1326007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3888 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org
2013-04-23 15:58:23 +00:00
parent d25b602dc0
commit ccd4b2aec8
11 changed files with 60 additions and 43 deletions
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13
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
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@@ -1224,9 +1224,9 @@ int32_t ModuleRtpRtcpImpl::ResetRTT(const uint32_t remote_ssrc) {
return rtcp_receiver_.ResetRTT(remote_ssrc);
}
void ModuleRtpRtcpImpl:: SetRtt(uint32_t rtt) {
void ModuleRtpRtcpImpl::SetRtt(uint32_t rtt) {
WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SetRtt(rtt: %u)", rtt);
rtcp_receiver_.SetRTT(static_cast<uint16_t>(rtt));
rtp_receiver_->set_rtt_ms(rtt);
}
// Reset RTP statistics.
@@ -1545,13 +1545,8 @@ int32_t ModuleRtpRtcpImpl::SendNACK(const uint16_t* nack_list,
id_,
"SendNACK(size:%u)", size);
uint16_t avg_rtt = 0;
rtcp_receiver_.RTT(rtp_receiver_->SSRC(), NULL, &avg_rtt, NULL, NULL);
int64_t wait_time = 5 + ((avg_rtt * 3) >> 1); // 5 + RTT * 1.5.
if (wait_time == 5) {
wait_time = 100; // During startup we don't have an RTT.
}
// 5 + RTT * 1.5.
int64_t wait_time = 5 + ((rtp_receiver_->rtt_ms() * 3) / 2);
const int64_t now = clock_->TimeInMilliseconds();
const int64_t time_limit = now - wait_time;
uint16_t nackLength = size;