| Index: webrtc/modules/rtp_rtcp/source/rtcp_sender.h
|
| diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
|
| index f59676685dbe81e27843953680f3b21ef155ff02..9c2a71499ad88152e3f9c18434387df98f34d413 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
|
| @@ -21,6 +21,7 @@
|
| #include "webrtc/api/call/transport.h"
|
| #include "webrtc/base/constructormagic.h"
|
| #include "webrtc/base/criticalsection.h"
|
| +#include "webrtc/base/optional.h"
|
| #include "webrtc/base/random.h"
|
| #include "webrtc/base/thread_annotations.h"
|
| #include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
|
| @@ -150,6 +151,7 @@ class RTCPSender {
|
| void SetCsrcs(const std::vector<uint32_t>& csrcs);
|
|
|
| void SetTargetBitrate(unsigned int target_bitrate);
|
| + void SetVideoBitrateAllocation(const BitrateAllocation& bitrate);
|
| bool SendFeedbackPacket(const rtcp::TransportFeedback& packet);
|
|
|
| private:
|
| @@ -197,6 +199,9 @@ class RTCPSender {
|
| EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
|
| std::unique_ptr<rtcp::RtcpPacket> BuildDlrr(const RtcpContext& context)
|
| EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
|
| + std::unique_ptr<rtcp::RtcpPacket> BuildTargetBitrate(
|
| + const RtcpContext& context)
|
| + EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
|
|
|
| private:
|
| const bool audio_;
|
| @@ -265,6 +270,9 @@ class RTCPSender {
|
|
|
| RTCPUtility::NackStats nack_stats_ GUARDED_BY(critical_section_rtcp_sender_);
|
|
|
| + rtc::Optional<BitrateAllocation> video_bitrate_allocation_
|
| + GUARDED_BY(critical_section_rtcp_sender_);
|
| +
|
| void SetFlag(RTCPPacketType type, bool is_volatile)
|
| EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
|
| void SetFlags(const std::set<RTCPPacketType>& types, bool is_volatile)
|
|
|