webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h - Issue 2007743003: Add sender controlled playout delay limits

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.h

Issue 2007743003: Add sender controlled playout delay limits (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@cleanup_rtp_hdr_extensions

Patch Set: Remove notion of max and current header length Created 4 years, 7 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

« webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc ('K') | « webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc ('k') | webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc » ('j') | webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc » ('J')
Expand Comments ('e') | Collapse Comments ('c') | Hide Comments ('s')

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

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

index 7bbb06e5289962031e33def3cc4da7f1222a995d..0f9ad74682e6a04a3f14c2a7ea75302a8f83523a 100644

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

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

@@ -19,6 +19,7 @@

#include "webrtc/base/criticalsection.h"

#include "webrtc/base/gtest_prod_util.h"

#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"

+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"

#include "webrtc/modules/rtp_rtcp/source/packet_loss_stats.h"

#include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"

#include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"

@@ -318,6 +319,7 @@ class ModuleRtpRtcpImpl : public RtpRtcp {

const override;

void OnReceivedNACK(const std::list<uint16_t>& nack_sequence_numbers);

+ void OnReceivedRtcpReceiverReport(const ReportBlockList& report_blocks);

void OnRequestSendReport();