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

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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: Removed audio changes and added locking on playout delay oracle Created 4 years, 7 months ago

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« webrtc/modules/rtp_rtcp/source/rtp_header_extension.h ('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.cc » ('J')
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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();