webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc - Issue 1669623004: Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt()

Keyboard Shortcuts

	File
u :	up to issue
j / k :	jump to file after / before current file
J / K :	jump to next file with a comment after / before current file
	Side-by-side diff
i :	toggle intra-line diffs
e :	expand all comments
c :	collapse all comments
s :	toggle showing all comments
n / p :	next / previous diff chunk or comment
N / P :	next / previous comment
<Up> / <Down> :	next / previous line

	Issue
u :	up to list of issues
j / k :	jump to patch after / before current patch
o / <Enter> :	open current patch in side-by-side view
i :	open current patch in unified diff view

	Issue List
j / k :	jump to issue after / before current issue
o / <Enter> :	open current issue

Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

Issue 1669623004: Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt() (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Addressed comment, rebase Created 4 years, 10 months ago

Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.

Jump to:

View side-by-side diff with in-line comments

Download patch

Index: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

index e6e2187de707ddac41f03854778707ac7363f18c..2c8786b906c4755f1222be95f6f2ed1b73217097 100644

--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

@@ -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.

« no previous file with comments | « webrtc/call/call.cc ('k') | webrtc/video/call_stats.h » ('j') | no next file with comments »