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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 <cstddef>
#include <cstdint>
#include <functional>
#include <map>
#include <memory>
#include <optional>
#include <set>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
#include "api/array_view.h"
#include "api/call/transport.h"
#include "api/environment/environment.h"
#include "api/rtp_headers.h"
#include "api/units/data_rate.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "api/video/video_bitrate_allocation.h"
#include "modules/rtp_rtcp/include/receive_statistics.h"
#include "modules/rtp_rtcp/include/rtcp_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/loss_notification.h"
#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
#include "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
#include "modules/rtp_rtcp/source/rtcp_receiver.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
#include "rtc_base/random.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
#include "system_wrappers/include/ntp_time.h"
namespace webrtc {
class RTCPSender final {
public:
struct Configuration {
// TODO(bugs.webrtc.org/11581): Remove this temporary conversion utility
// once rtc_rtcp_impl.cc/h are gone.
static Configuration FromRtpRtcpConfiguration(
const RtpRtcpInterface::Configuration& config);
// True for a audio version of the RTP/RTCP module object false will create
// a video version.
bool audio = false;
// SSRCs for media and retransmission, respectively.
// FlexFec SSRC is fetched from `flexfec_sender`.
uint32_t local_media_ssrc = 0;
// Transport object that will be called when packets are ready to be sent
// out on the network.
Transport* outgoing_transport = nullptr;
// Estimate RTT as non-sender as described in
// https://tools.ietf.org/html/rfc3611#section-4.4 and #section-4.5
bool non_sender_rtt_measurement = false;
// Optional callback which, if specified, is used by RTCPSender to schedule
// the next time to evaluate if RTCP should be sent by means of
// TimeToSendRTCPReport/SendRTCP.
// The RTCPSender client still needs to call TimeToSendRTCPReport/SendRTCP
// to actually get RTCP sent.
//
// Note: It's recommended to use the callback to ensure program design that
// doesn't use polling.
// TODO(bugs.webrtc.org/11581): Make mandatory once downstream consumers
// have migrated to the callback solution.
std::function<void(TimeDelta)> schedule_next_rtcp_send_evaluation_function;
std::optional<TimeDelta> rtcp_report_interval;
ReceiveStatisticsProvider* receive_statistics = nullptr;
RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer = nullptr;
};
struct FeedbackState {
FeedbackState();
FeedbackState(const FeedbackState&);
FeedbackState(FeedbackState&&);
~FeedbackState();
uint32_t packets_sent;
size_t media_bytes_sent;
DataRate send_bitrate;
uint32_t remote_sr;
NtpTime last_rr;
std::vector<rtcp::ReceiveTimeInfo> last_xr_rtis;
// Used when generating TMMBR.
RTCPReceiver* receiver;
};
RTCPSender(const Environment& env, Configuration config);
RTCPSender() = delete;
RTCPSender(const RTCPSender&) = delete;
RTCPSender& operator=(const RTCPSender&) = delete;
~RTCPSender();
RtcpMode Status() const RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
void SetRTCPStatus(RtcpMode method) RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
bool Sending() const RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
void SetSendingStatus(const FeedbackState& feedback_state,
bool enabled)
RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_); // combine the functions
void SetNonSenderRttMeasurement(bool enabled)
RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
void SetTimestampOffset(uint32_t timestamp_offset)
RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
void SetLastRtpTime(uint32_t rtp_timestamp,
std::optional<Timestamp> capture_time,
std::optional<int8_t> payload_type)
RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
void SetRtpClockRate(int8_t payload_type, int rtp_clock_rate_hz)
RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
uint32_t SSRC() const;
void SetSsrc(uint32_t ssrc);
void SetRemoteSSRC(uint32_t ssrc) RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
int32_t SetCNAME(absl::string_view cName)
RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
bool TimeToSendRTCPReport(bool send_keyframe_before_rtp = false) const
RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
int32_t SendRTCP(const FeedbackState& feedback_state,
RTCPPacketType packetType,
ArrayView<const uint16_t> nacks = {})
RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
int32_t SendLossNotification(const FeedbackState& feedback_state,
uint16_t last_decoded_seq_num,
uint16_t last_received_seq_num,
bool decodability_flag,
bool buffering_allowed)
RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs)
RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
void UnsetRemb() RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
bool TMMBR() const RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
void SetMaxRtpPacketSize(size_t max_packet_size)
RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
void SetTmmbn(std::vector<rtcp::TmmbItem> bounding_set)
RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
void SetCsrcs(const std::vector<uint32_t>& csrcs)
RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
void SetTargetBitrate(unsigned int target_bitrate)
RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
void SetVideoBitrateAllocation(const VideoBitrateAllocation& bitrate)
RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
void SendCombinedRtcpPacket(
std::vector<std::unique_ptr<rtcp::RtcpPacket>> rtcp_packets)
RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
private:
class RtcpContext;
class PacketSender;
std::optional<int32_t> ComputeCompoundRTCPPacket(
const FeedbackState& feedback_state,
RTCPPacketType packet_type,
ArrayView<const uint16_t> nacks,
PacketSender& sender) RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
TimeDelta ComputeTimeUntilNextReport(DataRate send_bitrate)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
// Determine which RTCP messages should be sent and setup flags.
void PrepareReport(const FeedbackState& feedback_state)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
std::vector<rtcp::ReportBlock> CreateReportBlocks(
const FeedbackState& feedback_state)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
void BuildSR(const RtcpContext& context, PacketSender& sender)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
void BuildRR(const RtcpContext& context, PacketSender& sender)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
void BuildSDES(const RtcpContext& context, PacketSender& sender)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
void BuildPLI(const RtcpContext& context, PacketSender& sender)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
void BuildREMB(const RtcpContext& context, PacketSender& sender)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
void BuildTMMBR(const RtcpContext& context, PacketSender& sender)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
void BuildTMMBN(const RtcpContext& context, PacketSender& sender)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
void BuildAPP(const RtcpContext& context, PacketSender& sender)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
void BuildLossNotification(const RtcpContext& context, PacketSender& sender)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
void BuildExtendedReports(const RtcpContext& context, PacketSender& sender)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
void BuildBYE(const RtcpContext& context, PacketSender& sender)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
void BuildFIR(const RtcpContext& context, PacketSender& sender)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
void BuildNACK(const RtcpContext& context, PacketSender& sender)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
// `duration` being TimeDelta::Zero() means schedule immediately.
void SetNextRtcpSendEvaluationDuration(TimeDelta duration)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
const Environment env_;
const bool audio_;
// TODO(bugs.webrtc.org/11581): `mutex_rtcp_sender_` shouldn't be required if
// we consistently run network related operations on the network thread.
// This is currently not possible due to callbacks from the process thread in
// ModuleRtpRtcpImpl2.
uint32_t ssrc_ RTC_GUARDED_BY(mutex_rtcp_sender_);
Random random_ RTC_GUARDED_BY(mutex_rtcp_sender_);
RtcpMode method_ RTC_GUARDED_BY(mutex_rtcp_sender_);
Transport* const transport_;
const TimeDelta report_interval_;
// Set from
// RTCPSender::Configuration::schedule_next_rtcp_send_evaluation_function.
const std::function<void(TimeDelta)>
schedule_next_rtcp_send_evaluation_function_;
mutable Mutex mutex_rtcp_sender_;
bool sending_ RTC_GUARDED_BY(mutex_rtcp_sender_);
std::optional<Timestamp> next_time_to_send_rtcp_
RTC_GUARDED_BY(mutex_rtcp_sender_);
uint32_t timestamp_offset_ RTC_GUARDED_BY(mutex_rtcp_sender_);
uint32_t last_rtp_timestamp_ RTC_GUARDED_BY(mutex_rtcp_sender_);
std::optional<Timestamp> last_frame_capture_time_
RTC_GUARDED_BY(mutex_rtcp_sender_);
// SSRC that we receive on our RTP channel
uint32_t remote_ssrc_ RTC_GUARDED_BY(mutex_rtcp_sender_);
std::string cname_ RTC_GUARDED_BY(mutex_rtcp_sender_);
ReceiveStatisticsProvider* receive_statistics_
RTC_GUARDED_BY(mutex_rtcp_sender_);
// send CSRCs
std::vector<uint32_t> csrcs_ RTC_GUARDED_BY(mutex_rtcp_sender_);
// Full intra request
uint8_t sequence_number_fir_ RTC_GUARDED_BY(mutex_rtcp_sender_);
rtcp::LossNotification loss_notification_ RTC_GUARDED_BY(mutex_rtcp_sender_);
// REMB
int64_t remb_bitrate_ RTC_GUARDED_BY(mutex_rtcp_sender_);
std::vector<uint32_t> remb_ssrcs_ RTC_GUARDED_BY(mutex_rtcp_sender_);
std::vector<rtcp::TmmbItem> tmmbn_to_send_ RTC_GUARDED_BY(mutex_rtcp_sender_);
uint32_t tmmbr_send_bps_ RTC_GUARDED_BY(mutex_rtcp_sender_);
uint32_t packet_oh_send_ RTC_GUARDED_BY(mutex_rtcp_sender_);
size_t max_packet_size_ RTC_GUARDED_BY(mutex_rtcp_sender_);
// True if sending of XR Receiver reference time report is enabled.
bool xr_send_receiver_reference_time_enabled_
RTC_GUARDED_BY(mutex_rtcp_sender_);
RtcpPacketTypeCounterObserver* const packet_type_counter_observer_;
RtcpPacketTypeCounter packet_type_counter_ RTC_GUARDED_BY(mutex_rtcp_sender_);
RtcpNackStats nack_stats_ RTC_GUARDED_BY(mutex_rtcp_sender_);
VideoBitrateAllocation video_bitrate_allocation_
RTC_GUARDED_BY(mutex_rtcp_sender_);
bool send_video_bitrate_allocation_ RTC_GUARDED_BY(mutex_rtcp_sender_);
std::map<int8_t, int> rtp_clock_rates_khz_ RTC_GUARDED_BY(mutex_rtcp_sender_);
int8_t last_payload_type_ RTC_GUARDED_BY(mutex_rtcp_sender_);
std::optional<VideoBitrateAllocation> CheckAndUpdateLayerStructure(
const VideoBitrateAllocation& bitrate) const
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
void SetFlag(uint32_t type, bool is_volatile)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
bool IsFlagPresent(uint32_t type) const
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
bool ConsumeFlag(uint32_t type, bool forced = false)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
bool AllVolatileFlagsConsumed() const
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_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(mutex_rtcp_sender_);
typedef void (RTCPSender::*BuilderFunc)(const RtcpContext&, PacketSender&);
// Map from RTCPPacketType to builder.
std::map<uint32_t, BuilderFunc> builders_;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_
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