media/cast/rtp_receiver/receiver_stats.cc - Issue 250363002: [Cast] Clean-up RtpCastHeader and RtpParser, removing the last WebRTC dependency.

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: media/cast/rtp_receiver/receiver_stats.cc

Issue 250363002: [Cast] Clean-up RtpCastHeader and RtpParser, removing the last WebRTC dependency. (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src

Patch Set: Created 6 years, 8 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

« no previous file with comments | « media/cast/framer/framer_unittest.cc ('k') | media/cast/rtp_receiver/receiver_stats_unittest.cc » ('j') | media/cast/rtp_receiver/rtp_parser/rtp_parser.h » ('J')
Expand Comments ('e') | Collapse Comments ('c') | Hide Comments ('s')

Index: media/cast/rtp_receiver/receiver_stats.cc

diff --git a/media/cast/rtp_receiver/receiver_stats.cc b/media/cast/rtp_receiver/receiver_stats.cc

index dd82991a3e969658762e6a3b4ffcbf49c9872ecd..7eff86763f88e8255177b41bdb3ac27f57035d72 100644

--- a/media/cast/rtp_receiver/receiver_stats.cc

+++ b/media/cast/rtp_receiver/receiver_stats.cc

@@ -75,7 +75,7 @@ void ReceiverStats::GetStatistics(uint8* fraction_lost,

}

void ReceiverStats::UpdateStatistics(const RtpCastHeader& header) {

- uint16 new_seq_num = header.webrtc.header.sequenceNumber;

+ const uint16 new_seq_num = header.sequence_number;

if (interval_number_packets_ == 0) {

// First packet in the interval.

@@ -99,7 +99,7 @@ void ReceiverStats::UpdateStatistics(const RtpCastHeader& header) {

// Compute Jitter.

base::TimeTicks now = clock_->NowTicks();

base::TimeDelta delta_new_timestamp =

- base::TimeDelta::FromMilliseconds(header.webrtc.header.timestamp);

+ base::TimeDelta::FromMilliseconds(header.rtp_timestamp);

if (total_number_packets_ > 0) {

// Update jitter.

base::TimeDelta delta =