zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt()

Browse Source 
Also move some stats reporting from vie_channel to send stats proxy

BUG=

Review URL: https://codereview.webrtc.org/1669623004

Cr-Commit-Position: refs/heads/master@{#11688}
This commit is contained in:
sprang
2016-02-19 09:03:26 -08:00
committed by Commit bot
parent 45c44f0b94
commit e2d83d6560
12 changed files with 92 additions and 141 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

@ -182,8 +182,12 @@ int32_t ModuleRtpRtcpImpl::Process() {
// Get processed rtt.
if (process_rtt) {
last_rtt_process_time_ = now;
if (rtt_stats_)
set_rtt_ms(rtt_stats_->LastProcessedRtt());
if (rtt_stats_) {
// Make sure we have a valid RTT before setting.
int64_t last_rtt = rtt_stats_->LastProcessedRtt();
if (last_rtt >= 0)
set_rtt_ms(last_rtt);
}
}
// For sending streams, make sure to not send a SR before media has been sent.