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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_
#define MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_
#include <map>
#include <memory>
#include <set>
#include <string>
#include <vector>
#include "absl/types/optional.h"
#include "api/call/transport.h"
#include "api/video/video_bitrate_allocation.h"
#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "modules/rtp_rtcp/include/receive_statistics.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtcp_nack_stats.h"
#include "modules/rtp_rtcp/source/rtcp_packet.h"
#include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
#include "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/random.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
class ModuleRtpRtcpImpl;
class RtcEventLog;
class RTCPSender {
public:
struct FeedbackState {
FeedbackState();
FeedbackState(const FeedbackState&);
FeedbackState(FeedbackState&&);
~FeedbackState();
uint32_t packets_sent;
size_t media_bytes_sent;
uint32_t send_bitrate;
uint32_t last_rr_ntp_secs;
uint32_t last_rr_ntp_frac;
uint32_t remote_sr;
std::vector<rtcp::ReceiveTimeInfo> last_xr_rtis;
// Used when generating TMMBR.
ModuleRtpRtcpImpl* module;
};
RTCPSender(bool audio,
Clock* clock,
ReceiveStatisticsProvider* receive_statistics,
RtcpPacketTypeCounterObserver* packet_type_counter_observer,
RtcEventLog* event_log,
Transport* outgoing_transport,
int report_interval_ms);
virtual ~RTCPSender();
RtcpMode Status() const;
void SetRTCPStatus(RtcpMode method);
bool Sending() const;
int32_t SetSendingStatus(const FeedbackState& feedback_state,
bool enabled); // combine the functions
int32_t SetNackStatus(bool enable);
void SetTimestampOffset(uint32_t timestamp_offset);
// TODO(bugs.webrtc.org/6458): Remove default parameter value when all the
// depending projects are updated to correctly set payload type.
void SetLastRtpTime(uint32_t rtp_timestamp,
int64_t capture_time_ms,
int8_t payload_type = -1);
void SetRtpClockRate(int8_t payload_type, int rtp_clock_rate_hz);
uint32_t SSRC() const;
void SetSSRC(uint32_t ssrc);
void SetRemoteSSRC(uint32_t ssrc);
int32_t SetCNAME(const char* cName);
int32_t AddMixedCNAME(uint32_t SSRC, const char* c_name);
int32_t RemoveMixedCNAME(uint32_t SSRC);
bool TimeToSendRTCPReport(bool sendKeyframeBeforeRTP = false) const;
int32_t SendRTCP(const FeedbackState& feedback_state,
RTCPPacketType packetType,
int32_t nackSize = 0,
const uint16_t* nackList = 0);
int32_t SendCompoundRTCP(const FeedbackState& feedback_state,
const std::set<RTCPPacketType>& packetTypes,
int32_t nackSize = 0,
const uint16_t* nackList = 0);
void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs);
void UnsetRemb();
bool TMMBR() const;
void SetTMMBRStatus(bool enable);
void SetMaxRtpPacketSize(size_t max_packet_size);
void SetTmmbn(std::vector<rtcp::TmmbItem> bounding_set);
int32_t SetApplicationSpecificData(uint8_t subType,
uint32_t name,
const uint8_t* data,
uint16_t length);
void SendRtcpXrReceiverReferenceTime(bool enable);
bool RtcpXrReceiverReferenceTime() const;
void SetCsrcs(const std::vector<uint32_t>& csrcs);
void SetTargetBitrate(unsigned int target_bitrate);
void SetVideoBitrateAllocation(const VideoBitrateAllocation& bitrate);
bool SendFeedbackPacket(const rtcp::TransportFeedback& packet);
private:
class RtcpContext;
// Determine which RTCP messages should be sent and setup flags.
void PrepareReport(const FeedbackState& feedback_state)
RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
std::vector<rtcp::ReportBlock> CreateReportBlocks(
const FeedbackState& feedback_state)
RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
std::unique_ptr<rtcp::RtcpPacket> BuildSR(const RtcpContext& context)
RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
std::unique_ptr<rtcp::RtcpPacket> BuildRR(const RtcpContext& context)
RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
std::unique_ptr<rtcp::RtcpPacket> BuildSDES(const RtcpContext& context)
RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
std::unique_ptr<rtcp::RtcpPacket> BuildPLI(const RtcpContext& context)
RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
std::unique_ptr<rtcp::RtcpPacket> BuildREMB(const RtcpContext& context)
RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
std::unique_ptr<rtcp::RtcpPacket> BuildTMMBR(const RtcpContext& context)
RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
std::unique_ptr<rtcp::RtcpPacket> BuildTMMBN(const RtcpContext& context)
RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
std::unique_ptr<rtcp::RtcpPacket> BuildAPP(const RtcpContext& context)
RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
std::unique_ptr<rtcp::RtcpPacket> BuildExtendedReports(
const RtcpContext& context)
RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
std::unique_ptr<rtcp::RtcpPacket> BuildBYE(const RtcpContext& context)
RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
std::unique_ptr<rtcp::RtcpPacket> BuildFIR(const RtcpContext& context)
RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
std::unique_ptr<rtcp::RtcpPacket> BuildNACK(const RtcpContext& context)
RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
private:
const bool audio_;
Clock* const clock_;
Random random_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
RtcpMode method_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
RtcEventLog* const event_log_;
Transport* const transport_;
const int report_interval_ms_;
rtc::CriticalSection critical_section_rtcp_sender_;
bool sending_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
int64_t next_time_to_send_rtcp_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
uint32_t timestamp_offset_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
uint32_t last_rtp_timestamp_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
int64_t last_frame_capture_time_ms_
RTC_GUARDED_BY(critical_section_rtcp_sender_);
uint32_t ssrc_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
// SSRC that we receive on our RTP channel
uint32_t remote_ssrc_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
std::string cname_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
ReceiveStatisticsProvider* receive_statistics_
RTC_GUARDED_BY(critical_section_rtcp_sender_);
std::map<uint32_t, std::string> csrc_cnames_
RTC_GUARDED_BY(critical_section_rtcp_sender_);
// send CSRCs
std::vector<uint32_t> csrcs_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
// Full intra request
uint8_t sequence_number_fir_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
// REMB
int64_t remb_bitrate_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
std::vector<uint32_t> remb_ssrcs_
RTC_GUARDED_BY(critical_section_rtcp_sender_);
std::vector<rtcp::TmmbItem> tmmbn_to_send_
RTC_GUARDED_BY(critical_section_rtcp_sender_);
uint32_t tmmbr_send_bps_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
uint32_t packet_oh_send_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
size_t max_packet_size_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
// APP
uint8_t app_sub_type_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
uint32_t app_name_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
std::unique_ptr<uint8_t[]> app_data_
RTC_GUARDED_BY(critical_section_rtcp_sender_);
uint16_t app_length_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
// True if sending of XR Receiver reference time report is enabled.
bool xr_send_receiver_reference_time_enabled_
RTC_GUARDED_BY(critical_section_rtcp_sender_);
RtcpPacketTypeCounterObserver* const packet_type_counter_observer_;
RtcpPacketTypeCounter packet_type_counter_
RTC_GUARDED_BY(critical_section_rtcp_sender_);
RtcpNackStats nack_stats_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
VideoBitrateAllocation video_bitrate_allocation_
RTC_GUARDED_BY(critical_section_rtcp_sender_);
bool send_video_bitrate_allocation_
RTC_GUARDED_BY(critical_section_rtcp_sender_);
std::map<int8_t, int> rtp_clock_rates_khz_
RTC_GUARDED_BY(critical_section_rtcp_sender_);
int8_t last_payload_type_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
absl::optional<VideoBitrateAllocation> CheckAndUpdateLayerStructure(
const VideoBitrateAllocation& bitrate) const
RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
void SetFlag(uint32_t type, bool is_volatile)
RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
void SetFlags(const std::set<RTCPPacketType>& types, bool is_volatile)
RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
bool IsFlagPresent(uint32_t type) const
RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
bool ConsumeFlag(uint32_t type, bool forced = false)
RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
bool AllVolatileFlagsConsumed() const
RTC_EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
struct ReportFlag {
ReportFlag(uint32_t type, bool is_volatile)
: type(type), is_volatile(is_volatile) {}
bool operator<(const ReportFlag& flag) const { return type < flag.type; }
bool operator==(const ReportFlag& flag) const { return type == flag.type; }
const uint32_t type;
const bool is_volatile;
};
std::set<ReportFlag> report_flags_
RTC_GUARDED_BY(critical_section_rtcp_sender_);
typedef std::unique_ptr<rtcp::RtcpPacket> (RTCPSender::*BuilderFunc)(
const RtcpContext&);
// Map from RTCPPacketType to builder.
std::map<uint32_t, BuilderFunc> builders_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTCPSender);
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_