mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 14:20:45 +01:00

* LossNotificationController is the class that decides when to issue LossNotification RTCP messages. * RtpRtcp handles the technicalities of producing RTCP messages. Bug: webrtc:10336 Change-Id: I292536257a984ca85d21d9cfa38e7ff2569cbb39 Reviewed-on: https://webrtc-review.googlesource.com/c/124123 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Elad Alon <eladalon@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26840}
306 lines
12 KiB
C++
306 lines
12 KiB
C++
/*
|
|
* 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);
|
|
|
|
int32_t SendLossNotification(const FeedbackState& feedback_state,
|
|
uint16_t last_decoded_seq_num,
|
|
uint16_t last_received_seq_num,
|
|
bool decodability_flag);
|
|
|
|
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> BuildLossNotification(
|
|
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_);
|
|
|
|
// Loss Notification
|
|
struct LossNotificationState {
|
|
uint16_t last_decoded_seq_num;
|
|
uint16_t last_received_seq_num;
|
|
bool decodability_flag;
|
|
};
|
|
LossNotificationState loss_notification_state_
|
|
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_
|