mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-12 21:30:45 +01:00

which needs to be added to the remote codecs a=fmtp: This also forces SimulcastCastEncoderAdapter to avoid issues with codecs that have native simulcast capability but do require synchronized keyframes. This parameter allows for large-scale experimentation and A/B testing whether the new behavior has advantages. It is to be considered transitional and may be removed again in the future. BUG=webrtc:10107 Change-Id: I81f496c987b2fed7ff3089efb746e7e89e89c033 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333560 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41805}
420 lines
15 KiB
C++
420 lines
15 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_RECEIVER_H_
|
|
#define MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_H_
|
|
|
|
#include <list>
|
|
#include <map>
|
|
#include <string>
|
|
#include <vector>
|
|
|
|
#include "absl/types/optional.h"
|
|
#include "api/array_view.h"
|
|
#include "api/sequence_checker.h"
|
|
#include "api/units/time_delta.h"
|
|
#include "api/units/timestamp.h"
|
|
#include "modules/rtp_rtcp/include/report_block_data.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/dlrr.h"
|
|
#include "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
|
|
#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
|
|
#include "rtc_base/containers/flat_map.h"
|
|
#include "rtc_base/synchronization/mutex.h"
|
|
#include "rtc_base/system/no_unique_address.h"
|
|
#include "rtc_base/thread_annotations.h"
|
|
#include "system_wrappers/include/ntp_time.h"
|
|
|
|
namespace webrtc {
|
|
|
|
class ModuleRtpRtcpImpl2;
|
|
class VideoBitrateAllocationObserver;
|
|
|
|
namespace rtcp {
|
|
class CommonHeader;
|
|
class ReportBlock;
|
|
class Rrtr;
|
|
class TargetBitrate;
|
|
class TmmbItem;
|
|
} // namespace rtcp
|
|
|
|
class RTCPReceiver final {
|
|
public:
|
|
class ModuleRtpRtcp {
|
|
public:
|
|
virtual void SetTmmbn(std::vector<rtcp::TmmbItem> bounding_set) = 0;
|
|
virtual void OnRequestSendReport() = 0;
|
|
virtual void OnReceivedNack(
|
|
const std::vector<uint16_t>& nack_sequence_numbers) = 0;
|
|
virtual void OnReceivedRtcpReportBlocks(
|
|
rtc::ArrayView<const ReportBlockData> report_blocks) = 0;
|
|
|
|
protected:
|
|
virtual ~ModuleRtpRtcp() = default;
|
|
};
|
|
// Standardized stats derived from the non-sender RTT.
|
|
class NonSenderRttStats {
|
|
public:
|
|
NonSenderRttStats() = default;
|
|
NonSenderRttStats(const NonSenderRttStats&) = default;
|
|
NonSenderRttStats& operator=(const NonSenderRttStats&) = default;
|
|
~NonSenderRttStats() = default;
|
|
void Update(TimeDelta non_sender_rtt_seconds) {
|
|
round_trip_time_ = non_sender_rtt_seconds;
|
|
total_round_trip_time_ += non_sender_rtt_seconds;
|
|
round_trip_time_measurements_++;
|
|
}
|
|
void Invalidate() { round_trip_time_.reset(); }
|
|
// https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
|
|
absl::optional<TimeDelta> round_trip_time() const {
|
|
return round_trip_time_;
|
|
}
|
|
// https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime
|
|
TimeDelta total_round_trip_time() const { return total_round_trip_time_; }
|
|
// https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements
|
|
int round_trip_time_measurements() const {
|
|
return round_trip_time_measurements_;
|
|
}
|
|
|
|
private:
|
|
absl::optional<TimeDelta> round_trip_time_;
|
|
TimeDelta total_round_trip_time_ = TimeDelta::Zero();
|
|
int round_trip_time_measurements_ = 0;
|
|
};
|
|
|
|
RTCPReceiver(const RtpRtcpInterface::Configuration& config,
|
|
ModuleRtpRtcp* owner);
|
|
|
|
RTCPReceiver(const RtpRtcpInterface::Configuration& config,
|
|
ModuleRtpRtcpImpl2* owner);
|
|
|
|
~RTCPReceiver();
|
|
|
|
void IncomingPacket(rtc::ArrayView<const uint8_t> packet);
|
|
|
|
int64_t LastReceivedReportBlockMs() const;
|
|
|
|
void set_local_media_ssrc(uint32_t ssrc);
|
|
uint32_t local_media_ssrc() const;
|
|
|
|
void SetRemoteSSRC(uint32_t ssrc);
|
|
uint32_t RemoteSSRC() const;
|
|
|
|
bool receiver_only() const { return receiver_only_; }
|
|
|
|
// Returns stats based on the received RTCP Sender Reports.
|
|
absl::optional<RtpRtcpInterface::SenderReportStats> GetSenderReportStats()
|
|
const;
|
|
|
|
std::vector<rtcp::ReceiveTimeInfo> ConsumeReceivedXrReferenceTimeInfo();
|
|
|
|
absl::optional<TimeDelta> AverageRtt() const;
|
|
absl::optional<TimeDelta> LastRtt() const;
|
|
|
|
// Returns non-sender RTT metrics for the remote SSRC.
|
|
NonSenderRttStats GetNonSenderRTT() const;
|
|
|
|
void SetNonSenderRttMeasurement(bool enabled);
|
|
absl::optional<TimeDelta> GetAndResetXrRrRtt();
|
|
|
|
// Called once per second on the worker thread to do rtt calculations.
|
|
// Returns an optional rtt value if one is available.
|
|
absl::optional<TimeDelta> OnPeriodicRttUpdate(Timestamp newer_than,
|
|
bool sending);
|
|
|
|
// A snapshot of Report Blocks with additional data of interest to statistics.
|
|
// Within this list, the source SSRC is unique and ReportBlockData represents
|
|
// the latest Report Block that was received for that SSRC.
|
|
std::vector<ReportBlockData> GetLatestReportBlockData() const;
|
|
|
|
// Returns true if we haven't received an RTCP RR for several RTCP
|
|
// intervals, but only triggers true once.
|
|
bool RtcpRrTimeout();
|
|
|
|
// Returns true if we haven't received an RTCP RR telling the receive side
|
|
// has not received RTP packets for too long, i.e. extended highest sequence
|
|
// number hasn't increased for several RTCP intervals. The function only
|
|
// returns true once until a new RR is received.
|
|
bool RtcpRrSequenceNumberTimeout();
|
|
|
|
std::vector<rtcp::TmmbItem> TmmbrReceived();
|
|
// Return true if new bandwidth should be set.
|
|
bool UpdateTmmbrTimers();
|
|
std::vector<rtcp::TmmbItem> BoundingSet(bool* tmmbr_owner);
|
|
// Set new bandwidth and notify remote clients about it.
|
|
void NotifyTmmbrUpdated();
|
|
|
|
private:
|
|
#if RTC_DCHECK_IS_ON
|
|
class CustomSequenceChecker : public SequenceChecker {
|
|
public:
|
|
explicit CustomSequenceChecker(bool disable_checks)
|
|
: disable_checks_(disable_checks) {}
|
|
bool IsCurrent() const {
|
|
if (disable_checks_)
|
|
return true;
|
|
return SequenceChecker::IsCurrent();
|
|
}
|
|
|
|
private:
|
|
const bool disable_checks_;
|
|
};
|
|
#else
|
|
class CustomSequenceChecker : public SequenceChecker {
|
|
public:
|
|
explicit CustomSequenceChecker(bool) {}
|
|
};
|
|
#endif
|
|
|
|
// A lightweight inlined set of local SSRCs.
|
|
class RegisteredSsrcs {
|
|
public:
|
|
static constexpr size_t kMediaSsrcIndex = 0;
|
|
// Initializes the set of registered local SSRCS by extracting them from the
|
|
// provided `config`. The `disable_sequence_checker` flag is a workaround
|
|
// to be able to use a sequence checker without breaking downstream
|
|
// code that currently doesn't follow the same threading rules as webrtc.
|
|
RegisteredSsrcs(bool disable_sequence_checker,
|
|
const RtpRtcpInterface::Configuration& config);
|
|
|
|
// Indicates if `ssrc` is in the set of registered local SSRCs.
|
|
bool contains(uint32_t ssrc) const;
|
|
uint32_t media_ssrc() const;
|
|
void set_media_ssrc(uint32_t ssrc);
|
|
|
|
private:
|
|
RTC_NO_UNIQUE_ADDRESS CustomSequenceChecker packet_sequence_checker_;
|
|
absl::InlinedVector<uint32_t, kMaxSimulcastStreams> ssrcs_
|
|
RTC_GUARDED_BY(packet_sequence_checker_);
|
|
};
|
|
|
|
struct PacketInformation;
|
|
|
|
// Structure for handing TMMBR and TMMBN rtcp messages (RFC5104,
|
|
// section 3.5.4).
|
|
struct TmmbrInformation {
|
|
struct TimedTmmbrItem {
|
|
rtcp::TmmbItem tmmbr_item;
|
|
Timestamp last_updated = Timestamp::Zero();
|
|
};
|
|
|
|
Timestamp last_time_received = Timestamp::Zero();
|
|
|
|
bool ready_for_delete = false;
|
|
|
|
std::vector<rtcp::TmmbItem> tmmbn;
|
|
std::map<uint32_t, TimedTmmbrItem> tmmbr;
|
|
};
|
|
|
|
// Structure for storing received RRTR RTCP messages (RFC3611, section 4.4).
|
|
struct RrtrInformation {
|
|
RrtrInformation(uint32_t ssrc,
|
|
uint32_t received_remote_mid_ntp_time,
|
|
uint32_t local_receive_mid_ntp_time)
|
|
: ssrc(ssrc),
|
|
received_remote_mid_ntp_time(received_remote_mid_ntp_time),
|
|
local_receive_mid_ntp_time(local_receive_mid_ntp_time) {}
|
|
|
|
uint32_t ssrc;
|
|
// Received NTP timestamp in compact representation.
|
|
uint32_t received_remote_mid_ntp_time;
|
|
// NTP time when the report was received in compact representation.
|
|
uint32_t local_receive_mid_ntp_time;
|
|
};
|
|
|
|
struct LastFirStatus {
|
|
LastFirStatus(Timestamp now, uint8_t sequence_number)
|
|
: request(now), sequence_number(sequence_number) {}
|
|
Timestamp request;
|
|
uint8_t sequence_number;
|
|
};
|
|
|
|
class RttStats {
|
|
public:
|
|
RttStats() = default;
|
|
RttStats(const RttStats&) = default;
|
|
RttStats& operator=(const RttStats&) = default;
|
|
|
|
void AddRtt(TimeDelta rtt);
|
|
|
|
TimeDelta last_rtt() const { return last_rtt_; }
|
|
TimeDelta average_rtt() const { return sum_rtt_ / num_rtts_; }
|
|
|
|
private:
|
|
TimeDelta last_rtt_ = TimeDelta::Zero();
|
|
TimeDelta sum_rtt_ = TimeDelta::Zero();
|
|
size_t num_rtts_ = 0;
|
|
};
|
|
|
|
bool ParseCompoundPacket(rtc::ArrayView<const uint8_t> packet,
|
|
PacketInformation* packet_information);
|
|
|
|
void TriggerCallbacksFromRtcpPacket(
|
|
const PacketInformation& packet_information);
|
|
|
|
TmmbrInformation* FindOrCreateTmmbrInfo(uint32_t remote_ssrc)
|
|
RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
|
|
// Update TmmbrInformation (if present) is alive.
|
|
void UpdateTmmbrRemoteIsAlive(uint32_t remote_ssrc)
|
|
RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
|
|
TmmbrInformation* GetTmmbrInformation(uint32_t remote_ssrc)
|
|
RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
|
|
|
|
bool HandleSenderReport(const rtcp::CommonHeader& rtcp_block,
|
|
PacketInformation* packet_information)
|
|
RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
|
|
|
|
bool HandleReceiverReport(const rtcp::CommonHeader& rtcp_block,
|
|
PacketInformation* packet_information)
|
|
RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
|
|
|
|
void HandleReportBlock(const rtcp::ReportBlock& report_block,
|
|
PacketInformation* packet_information,
|
|
uint32_t remote_ssrc)
|
|
RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
|
|
|
|
bool HandleSdes(const rtcp::CommonHeader& rtcp_block,
|
|
PacketInformation* packet_information)
|
|
RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
|
|
|
|
bool HandleXr(const rtcp::CommonHeader& rtcp_block,
|
|
PacketInformation* packet_information,
|
|
bool& contains_dlrr,
|
|
uint32_t& ssrc)
|
|
RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
|
|
|
|
void HandleXrReceiveReferenceTime(uint32_t sender_ssrc,
|
|
const rtcp::Rrtr& rrtr)
|
|
RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
|
|
|
|
void HandleXrDlrrReportBlock(uint32_t ssrc, const rtcp::ReceiveTimeInfo& rti)
|
|
RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
|
|
|
|
void HandleXrTargetBitrate(uint32_t ssrc,
|
|
const rtcp::TargetBitrate& target_bitrate,
|
|
PacketInformation* packet_information)
|
|
RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
|
|
|
|
bool HandleNack(const rtcp::CommonHeader& rtcp_block,
|
|
PacketInformation* packet_information)
|
|
RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
|
|
|
|
bool HandleApp(const rtcp::CommonHeader& rtcp_block,
|
|
PacketInformation* packet_information)
|
|
RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
|
|
|
|
bool HandleBye(const rtcp::CommonHeader& rtcp_block)
|
|
RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
|
|
|
|
bool HandlePli(const rtcp::CommonHeader& rtcp_block,
|
|
PacketInformation* packet_information)
|
|
RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
|
|
|
|
void HandlePsfbApp(const rtcp::CommonHeader& rtcp_block,
|
|
PacketInformation* packet_information)
|
|
RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
|
|
|
|
bool HandleTmmbr(const rtcp::CommonHeader& rtcp_block,
|
|
PacketInformation* packet_information)
|
|
RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
|
|
|
|
bool HandleTmmbn(const rtcp::CommonHeader& rtcp_block,
|
|
PacketInformation* packet_information)
|
|
RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
|
|
|
|
bool HandleSrReq(const rtcp::CommonHeader& rtcp_block,
|
|
PacketInformation* packet_information)
|
|
RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
|
|
|
|
bool HandleFir(const rtcp::CommonHeader& rtcp_block,
|
|
PacketInformation* packet_information)
|
|
RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
|
|
|
|
void HandleTransportFeedback(const rtcp::CommonHeader& rtcp_block,
|
|
PacketInformation* packet_information)
|
|
RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
|
|
|
|
bool RtcpRrTimeoutLocked(Timestamp now)
|
|
RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
|
|
|
|
bool RtcpRrSequenceNumberTimeoutLocked(Timestamp now)
|
|
RTC_EXCLUSIVE_LOCKS_REQUIRED(rtcp_receiver_lock_);
|
|
|
|
Clock* const clock_;
|
|
const bool receiver_only_;
|
|
ModuleRtpRtcp* const rtp_rtcp_;
|
|
// The set of registered local SSRCs.
|
|
RegisteredSsrcs registered_ssrcs_;
|
|
|
|
NetworkLinkRtcpObserver* const network_link_rtcp_observer_;
|
|
RtcpIntraFrameObserver* const rtcp_intra_frame_observer_;
|
|
RtcpLossNotificationObserver* const rtcp_loss_notification_observer_;
|
|
NetworkStateEstimateObserver* const network_state_estimate_observer_;
|
|
VideoBitrateAllocationObserver* const bitrate_allocation_observer_;
|
|
const TimeDelta report_interval_;
|
|
|
|
mutable Mutex rtcp_receiver_lock_;
|
|
uint32_t remote_ssrc_ RTC_GUARDED_BY(rtcp_receiver_lock_);
|
|
|
|
// Received sender report.
|
|
RtpRtcpInterface::SenderReportStats remote_sender_
|
|
RTC_GUARDED_BY(rtcp_receiver_lock_);
|
|
|
|
// Received RRTR information in ascending receive time order.
|
|
std::list<RrtrInformation> received_rrtrs_
|
|
RTC_GUARDED_BY(rtcp_receiver_lock_);
|
|
// Received RRTR information mapped by remote ssrc.
|
|
flat_map<uint32_t, std::list<RrtrInformation>::iterator>
|
|
received_rrtrs_ssrc_it_ RTC_GUARDED_BY(rtcp_receiver_lock_);
|
|
|
|
// Estimated rtt, nullopt when there is no valid estimate.
|
|
bool xr_rrtr_status_ RTC_GUARDED_BY(rtcp_receiver_lock_);
|
|
absl::optional<TimeDelta> xr_rr_rtt_;
|
|
|
|
Timestamp oldest_tmmbr_info_ RTC_GUARDED_BY(rtcp_receiver_lock_);
|
|
// Mapped by remote ssrc.
|
|
flat_map<uint32_t, TmmbrInformation> tmmbr_infos_
|
|
RTC_GUARDED_BY(rtcp_receiver_lock_);
|
|
|
|
// Round-Trip Time per remote sender ssrc.
|
|
flat_map<uint32_t, RttStats> rtts_ RTC_GUARDED_BY(rtcp_receiver_lock_);
|
|
// Non-sender Round-trip time per remote ssrc.
|
|
flat_map<uint32_t, NonSenderRttStats> non_sender_rtts_
|
|
RTC_GUARDED_BY(rtcp_receiver_lock_);
|
|
|
|
// Report blocks per local source ssrc.
|
|
flat_map<uint32_t, ReportBlockData> received_report_blocks_
|
|
RTC_GUARDED_BY(rtcp_receiver_lock_);
|
|
flat_map<uint32_t, LastFirStatus> last_fir_
|
|
RTC_GUARDED_BY(rtcp_receiver_lock_);
|
|
|
|
// The last time we received an RTCP Report block for this module.
|
|
Timestamp last_received_rb_ RTC_GUARDED_BY(rtcp_receiver_lock_) =
|
|
Timestamp::PlusInfinity();
|
|
|
|
// The time we last received an RTCP RR telling we have successfully
|
|
// delivered RTP packet to the remote side.
|
|
Timestamp last_increased_sequence_number_ = Timestamp::PlusInfinity();
|
|
|
|
RtcpCnameCallback* const cname_callback_;
|
|
ReportBlockDataObserver* const report_block_data_observer_;
|
|
|
|
RtcpPacketTypeCounterObserver* const packet_type_counter_observer_;
|
|
RtcpPacketTypeCounter packet_type_counter_;
|
|
|
|
RtcpNackStats nack_stats_;
|
|
|
|
size_t num_skipped_packets_;
|
|
Timestamp last_skipped_packets_warning_;
|
|
};
|
|
} // namespace webrtc
|
|
#endif // MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_H_
|