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Use the RTT from RtcpRttStats class if provided when sending/receiving NACK.

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R=holmer@google.com

Review URL: https://webrtc-codereview.appspot.com/5049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5228 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
asapersson@webrtc.org
2013-12-05 15:06:56 +00:00
parent ebad765ee0
commit 96a9b2dcdc
3 changed files with 20 additions and 7 deletions
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18
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
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@ -1135,10 +1135,13 @@ int32_t ModuleRtpRtcpImpl::SendNACK(const uint16_t* nack_list,
id_,
"SendNACK(size:%u)", size);
uint16_t avg_rtt = 0;
rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), NULL, &avg_rtt, NULL, NULL);
// Use RTT from RtcpRttStats class if provided.
uint16_t rtt = rtt_ms();
if (rtt == 0) {
rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), NULL, &rtt, NULL, NULL);
}
int64_t wait_time = 5 + ((avg_rtt * 3) >> 1); // 5 + RTT * 1.5.
int64_t wait_time = 5 + ((rtt * 3) >> 1); // 5 + RTT * 1.5.
if (wait_time == 5) {
wait_time = 100; // During startup we don't have an RTT.
}
@ -1597,9 +1600,12 @@ void ModuleRtpRtcpImpl::OnReceivedNACK(
nack_sequence_numbers.size() == 0) {
return;
}
uint16_t avg_rtt = 0;
rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), NULL, &avg_rtt, NULL, NULL);
rtp_sender_.OnReceivedNACK(nack_sequence_numbers, avg_rtt);
// Use RTT from RtcpRttStats class if provided.
uint16_t rtt = rtt_ms();
if (rtt == 0) {
rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), NULL, &rtt, NULL, NULL);
}
rtp_sender_.OnReceivedNACK(nack_sequence_numbers, rtt);
}
int32_t ModuleRtpRtcpImpl::LastReceivedNTP(