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webrtc / src / 71604ebf6425a62331505e5d14b1f7af4095074a / . / modules / rtp_rtcp / source / rtcp_receiver.cc
blob: 47843be810f53a462e3afb7017f9b4b21ae573d9 [file] [log] [blame]
/*
* 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.
*/
#include "modules/rtp_rtcp/source/rtcp_receiver.h"
#include <string.h>
#include <algorithm>
#include <limits>
#include <map>
#include <memory>
#include <utility>
#include <vector>
#include "api/video/video_bitrate_allocation.h"
#include "api/video/video_bitrate_allocator.h"
#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
#include "modules/rtp_rtcp/source/rtcp_packet/compound_packet.h"
#include "modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
#include "modules/rtp_rtcp/source/rtcp_packet/fir.h"
#include "modules/rtp_rtcp/source/rtcp_packet/loss_notification.h"
#include "modules/rtp_rtcp/source/rtcp_packet/nack.h"
#include "modules/rtp_rtcp/source/rtcp_packet/pli.h"
#include "modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
#include "modules/rtp_rtcp/source/rtcp_packet/remb.h"
#include "modules/rtp_rtcp/source/rtcp_packet/remote_estimate.h"
#include "modules/rtp_rtcp/source/rtcp_packet/sdes.h"
#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
#include "modules/rtp_rtcp/source/rtcp_packet/tmmbn.h"
#include "modules/rtp_rtcp/source/rtcp_packet/tmmbr.h"
#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
#include "modules/rtp_rtcp/source/time_util.h"
#include "modules/rtp_rtcp/source/tmmbr_help.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/ntp_time.h"
namespace webrtc {
namespace {
using rtcp::CommonHeader;
using rtcp::ReportBlock;
// The number of RTCP time intervals needed to trigger a timeout.
const int kRrTimeoutIntervals = 3;
const int64_t kTmmbrTimeoutIntervalMs = 5 * 5000;
const int64_t kMaxWarningLogIntervalMs = 10000;
const int64_t kRtcpMinFrameLengthMs = 17;
// Maximum number of received RRTRs that will be stored.
const size_t kMaxNumberOfStoredRrtrs = 300;
constexpr TimeDelta kDefaultVideoReportInterval = TimeDelta::Seconds(1);
constexpr TimeDelta kDefaultAudioReportInterval = TimeDelta::Seconds(5);
// Returns true if the `timestamp` has exceeded the |interval *
// kRrTimeoutIntervals| period and was reset (set to PlusInfinity()). Returns
// false if the timer was either already reset or if it has not expired.
bool ResetTimestampIfExpired(const Timestamp now,
Timestamp& timestamp,
TimeDelta interval) {
if (timestamp.IsInfinite() ||
now <= timestamp + interval * kRrTimeoutIntervals) {
return false;
}
timestamp = Timestamp::PlusInfinity();
return true;
}
} // namespace
constexpr size_t RTCPReceiver::RegisteredSsrcs::kMediaSsrcIndex;
constexpr size_t RTCPReceiver::RegisteredSsrcs::kMaxSsrcs;
RTCPReceiver::RegisteredSsrcs::RegisteredSsrcs(
bool disable_sequence_checker,
const RtpRtcpInterface::Configuration& config)
: packet_sequence_checker_(disable_sequence_checker) {
packet_sequence_checker_.Detach();
ssrcs_.push_back(config.local_media_ssrc);
if (config.rtx_send_ssrc) {
ssrcs_.push_back(*config.rtx_send_ssrc);
}
if (config.fec_generator) {
absl::optional<uint32_t> flexfec_ssrc = config.fec_generator->FecSsrc();
if (flexfec_ssrc) {
ssrcs_.push_back(*flexfec_ssrc);
}
}
// Ensure that the RegisteredSsrcs can inline the SSRCs.
RTC_DCHECK_LE(ssrcs_.size(), RTCPReceiver::RegisteredSsrcs::kMaxSsrcs);
}
bool RTCPReceiver::RegisteredSsrcs::contains(uint32_t ssrc) const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
return absl::c_linear_search(ssrcs_, ssrc);
}
uint32_t RTCPReceiver::RegisteredSsrcs::media_ssrc() const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
return ssrcs_[kMediaSsrcIndex];
}
void RTCPReceiver::RegisteredSsrcs::set_media_ssrc(uint32_t ssrc) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
ssrcs_[kMediaSsrcIndex] = ssrc;
}
struct RTCPReceiver::PacketInformation {
uint32_t packet_type_flags = 0; // RTCPPacketTypeFlags bit field.
uint32_t remote_ssrc = 0;
std::vector<uint16_t> nack_sequence_numbers;
// TODO(hbos): Remove `report_blocks` in favor of `report_block_datas`.
ReportBlockList report_blocks;
std::vector<ReportBlockData> report_block_datas;
int64_t rtt_ms = 0;
uint32_t receiver_estimated_max_bitrate_bps = 0;
std::unique_ptr<rtcp::TransportFeedback> transport_feedback;
absl::optional<VideoBitrateAllocation> target_bitrate_allocation;
absl::optional<NetworkStateEstimate> network_state_estimate;
std::unique_ptr<rtcp::LossNotification> loss_notification;
};
RTCPReceiver::RTCPReceiver(const RtpRtcpInterface::Configuration& config,
ModuleRtpRtcpImpl2* owner)
: clock_(config.clock),
receiver_only_(config.receiver_only),
rtp_rtcp_(owner),
main_ssrc_(config.local_media_ssrc),
registered_ssrcs_(false, config),
rtcp_bandwidth_observer_(config.bandwidth_callback),
rtcp_intra_frame_observer_(config.intra_frame_callback),
rtcp_loss_notification_observer_(config.rtcp_loss_notification_observer),
network_state_estimate_observer_(config.network_state_estimate_observer),
transport_feedback_observer_(config.transport_feedback_callback),
bitrate_allocation_observer_(config.bitrate_allocation_observer),
report_interval_(config.rtcp_report_interval_ms > 0
? TimeDelta::Millis(config.rtcp_report_interval_ms)
: (config.audio ? kDefaultAudioReportInterval
: kDefaultVideoReportInterval)),
// TODO(bugs.webrtc.org/10774): Remove fallback.
remote_ssrc_(0),
remote_sender_rtp_time_(0),
remote_sender_packet_count_(0),
remote_sender_octet_count_(0),
remote_sender_reports_count_(0),
xr_rrtr_status_(config.non_sender_rtt_measurement),
xr_rr_rtt_ms_(0),
oldest_tmmbr_info_ms_(0),
cname_callback_(config.rtcp_cname_callback),
report_block_data_observer_(config.report_block_data_observer),
packet_type_counter_observer_(config.rtcp_packet_type_counter_observer),
num_skipped_packets_(0),
last_skipped_packets_warning_ms_(clock_->TimeInMilliseconds()) {
RTC_DCHECK(owner);
}
RTCPReceiver::RTCPReceiver(const RtpRtcpInterface::Configuration& config,
ModuleRtpRtcp* owner)
: clock_(config.clock),
receiver_only_(config.receiver_only),
rtp_rtcp_(owner),
main_ssrc_(config.local_media_ssrc),
registered_ssrcs_(true, config),
rtcp_bandwidth_observer_(config.bandwidth_callback),
rtcp_intra_frame_observer_(config.intra_frame_callback),
rtcp_loss_notification_observer_(config.rtcp_loss_notification_observer),
network_state_estimate_observer_(config.network_state_estimate_observer),
transport_feedback_observer_(config.transport_feedback_callback),
bitrate_allocation_observer_(config.bitrate_allocation_observer),
report_interval_(config.rtcp_report_interval_ms > 0
? TimeDelta::Millis(config.rtcp_report_interval_ms)
: (config.audio ? kDefaultAudioReportInterval
: kDefaultVideoReportInterval)),
// TODO(bugs.webrtc.org/10774): Remove fallback.
remote_ssrc_(0),
remote_sender_rtp_time_(0),
remote_sender_packet_count_(0),
remote_sender_octet_count_(0),
remote_sender_reports_count_(0),
xr_rrtr_status_(config.non_sender_rtt_measurement),
xr_rr_rtt_ms_(0),
oldest_tmmbr_info_ms_(0),
cname_callback_(config.rtcp_cname_callback),
report_block_data_observer_(config.report_block_data_observer),
packet_type_counter_observer_(config.rtcp_packet_type_counter_observer),
num_skipped_packets_(0),
last_skipped_packets_warning_ms_(clock_->TimeInMilliseconds()) {
RTC_DCHECK(owner);
// Dear reader - if you're here because of this log statement and are
// wondering what this is about, chances are that you are using an instance
// of RTCPReceiver without using the webrtc APIs. This creates a bit of a
// problem for WebRTC because this class is a part of an internal
// implementation that is constantly changing and being improved.
// The intention of this log statement is to give a heads up that changes
// are coming and encourage you to use the public APIs or be prepared that
// things might break down the line as more changes land. A thing you could
// try out for now is to replace the `CustomSequenceChecker` in the header
// with a regular `SequenceChecker` and see if that triggers an
// error in your code. If it does, chances are you have your own threading
// model that is not the same as WebRTC internally has.
RTC_LOG(LS_INFO) << "************** !!!DEPRECATION WARNING!! **************";
}
RTCPReceiver::~RTCPReceiver() {}
void RTCPReceiver::IncomingPacket(rtc::ArrayView<const uint8_t> packet) {
if (packet.empty()) {
RTC_LOG(LS_WARNING) << "Incoming empty RTCP packet";
return;
}
PacketInformation packet_information;
if (!ParseCompoundPacket(packet, &packet_information))
return;
TriggerCallbacksFromRtcpPacket(packet_information);
}
// This method is only used by test and legacy code, so we should be able to
// remove it soon.
int64_t RTCPReceiver::LastReceivedReportBlockMs() const {
MutexLock lock(&rtcp_receiver_lock_);
return last_received_rb_.IsFinite() ? last_received_rb_.ms() : 0;
}
void RTCPReceiver::SetRemoteSSRC(uint32_t ssrc) {
MutexLock lock(&rtcp_receiver_lock_);
// New SSRC reset old reports.
last_received_sr_ntp_.Reset();
remote_ssrc_ = ssrc;
}
void RTCPReceiver::set_local_media_ssrc(uint32_t ssrc) {
registered_ssrcs_.set_media_ssrc(ssrc);
}
uint32_t RTCPReceiver::local_media_ssrc() const {
return registered_ssrcs_.media_ssrc();
}
uint32_t RTCPReceiver::RemoteSSRC() const {
MutexLock lock(&rtcp_receiver_lock_);
return remote_ssrc_;
}
void RTCPReceiver::RttStats::AddRtt(TimeDelta rtt) {
last_rtt_ = rtt;
if (rtt < min_rtt_) {
min_rtt_ = rtt;
}
if (rtt > max_rtt_) {
max_rtt_ = rtt;
}
sum_rtt_ += rtt;
++num_rtts_;
}
int32_t RTCPReceiver::RTT(uint32_t remote_ssrc,
int64_t* last_rtt_ms,
int64_t* avg_rtt_ms,
int64_t* min_rtt_ms,
int64_t* max_rtt_ms) const {
MutexLock lock(&rtcp_receiver_lock_);
auto it = rtts_.find(remote_ssrc);
if (it == rtts_.end()) {
return -1;
}
if (last_rtt_ms) {
*last_rtt_ms = it->second.last_rtt().ms();
}
if (avg_rtt_ms) {
*avg_rtt_ms = it->second.average_rtt().ms();
}
if (min_rtt_ms) {
*min_rtt_ms = it->second.min_rtt().ms();
}
if (max_rtt_ms) {
*max_rtt_ms = it->second.max_rtt().ms();
}
return 0;
}
RTCPReceiver::NonSenderRttStats RTCPReceiver::GetNonSenderRTT() const {
MutexLock lock(&rtcp_receiver_lock_);
auto it = non_sender_rtts_.find(remote_ssrc_);
if (it == non_sender_rtts_.end()) {
return {};
}
return it->second;
}
void RTCPReceiver::SetNonSenderRttMeasurement(bool enabled) {
MutexLock lock(&rtcp_receiver_lock_);
xr_rrtr_status_ = enabled;
}
bool RTCPReceiver::GetAndResetXrRrRtt(int64_t* rtt_ms) {
RTC_DCHECK(rtt_ms);
MutexLock lock(&rtcp_receiver_lock_);
if (xr_rr_rtt_ms_ == 0) {
return false;
}
*rtt_ms = xr_rr_rtt_ms_;
xr_rr_rtt_ms_ = 0;
return true;
}
// Called regularly (1/sec) on the worker thread to do rtt calculations.
absl::optional<TimeDelta> RTCPReceiver::OnPeriodicRttUpdate(
Timestamp newer_than,
bool sending) {
// Running on the worker thread (same as construction thread).
absl::optional<TimeDelta> rtt;
if (sending) {
// Check if we've received a report block within the last kRttUpdateInterval
// amount of time.
MutexLock lock(&rtcp_receiver_lock_);
if (last_received_rb_.IsInfinite() || last_received_rb_ > newer_than) {
TimeDelta max_rtt = TimeDelta::MinusInfinity();
for (const auto& rtt_stats : rtts_) {
if (rtt_stats.second.last_rtt() > max_rtt) {
max_rtt = rtt_stats.second.last_rtt();
}
}
if (max_rtt.IsFinite()) {
rtt = max_rtt;
}
}
// Check for expired timers and if so, log and reset.
auto now = clock_->CurrentTime();
if (RtcpRrTimeoutLocked(now)) {
RTC_LOG_F(LS_WARNING) << "Timeout: No RTCP RR received.";
} else if (RtcpRrSequenceNumberTimeoutLocked(now)) {
RTC_LOG_F(LS_WARNING) << "Timeout: No increase in RTCP RR extended "
"highest sequence number.";
}
} else {
// Report rtt from receiver.
int64_t rtt_ms;
if (GetAndResetXrRrRtt(&rtt_ms)) {
rtt.emplace(TimeDelta::Millis(rtt_ms));
}
}
return rtt;
}
bool RTCPReceiver::NTP(uint32_t* received_ntp_secs,
uint32_t* received_ntp_frac,
uint32_t* rtcp_arrival_time_secs,
uint32_t* rtcp_arrival_time_frac,
uint32_t* rtcp_timestamp,
uint32_t* remote_sender_packet_count,
uint64_t* remote_sender_octet_count,
uint64_t* remote_sender_reports_count) const {
MutexLock lock(&rtcp_receiver_lock_);
if (!last_received_sr_ntp_.Valid())
return false;
// NTP from incoming SenderReport.
if (received_ntp_secs)
*received_ntp_secs = remote_sender_ntp_time_.seconds();
if (received_ntp_frac)
*received_ntp_frac = remote_sender_ntp_time_.fractions();
// Rtp time from incoming SenderReport.
if (rtcp_timestamp)
*rtcp_timestamp = remote_sender_rtp_time_;
// Local NTP time when we received a RTCP packet with a send block.
if (rtcp_arrival_time_secs)
*rtcp_arrival_time_secs = last_received_sr_ntp_.seconds();
if (rtcp_arrival_time_frac)
*rtcp_arrival_time_frac = last_received_sr_ntp_.fractions();
// Counters.
if (remote_sender_packet_count)
*remote_sender_packet_count = remote_sender_packet_count_;
if (remote_sender_octet_count)
*remote_sender_octet_count = remote_sender_octet_count_;
if (remote_sender_reports_count)
*remote_sender_reports_count = remote_sender_reports_count_;
return true;
}
std::vector<rtcp::ReceiveTimeInfo>
RTCPReceiver::ConsumeReceivedXrReferenceTimeInfo() {
MutexLock lock(&rtcp_receiver_lock_);
const size_t last_xr_rtis_size = std::min(
received_rrtrs_.size(), rtcp::ExtendedReports::kMaxNumberOfDlrrItems);
std::vector<rtcp::ReceiveTimeInfo> last_xr_rtis;
last_xr_rtis.reserve(last_xr_rtis_size);
const uint32_t now_ntp = CompactNtp(clock_->CurrentNtpTime());
for (size_t i = 0; i < last_xr_rtis_size; ++i) {
RrtrInformation& rrtr = received_rrtrs_.front();
last_xr_rtis.emplace_back(rrtr.ssrc, rrtr.received_remote_mid_ntp_time,
now_ntp - rrtr.local_receive_mid_ntp_time);
received_rrtrs_ssrc_it_.erase(rrtr.ssrc);
received_rrtrs_.pop_front();
}
return last_xr_rtis;
}
std::vector<ReportBlockData> RTCPReceiver::GetLatestReportBlockData() const {
std::vector<ReportBlockData> result;
MutexLock lock(&rtcp_receiver_lock_);
for (const auto& report : received_report_blocks_) {
result.push_back(report.second);
}
return result;
}
bool RTCPReceiver::ParseCompoundPacket(rtc::ArrayView<const uint8_t> packet,
PacketInformation* packet_information) {
MutexLock lock(&rtcp_receiver_lock_);
CommonHeader rtcp_block;
// If a sender report is received but no DLRR, we need to reset the
// roundTripTime stat according to the standard, see
// https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
struct RtcpReceivedBlock {
bool sender_report = false;
bool dlrr = false;
};
// For each remote SSRC we store if we've received a sender report or a DLRR
// block.
flat_map<uint32_t, RtcpReceivedBlock> received_blocks;
for (const uint8_t* next_block = packet.begin(); next_block != packet.end();
next_block = rtcp_block.NextPacket()) {
ptrdiff_t remaining_blocks_size = packet.end() - next_block;
RTC_DCHECK_GT(remaining_blocks_size, 0);
if (!rtcp_block.Parse(next_block, remaining_blocks_size)) {
if (next_block == packet.begin()) {
// Failed to parse 1st header, nothing was extracted from this packet.
RTC_LOG(LS_WARNING) << "Incoming invalid RTCP packet";
return false;
}
++num_skipped_packets_;
break;
}
if (packet_type_counter_.first_packet_time_ms == -1)
packet_type_counter_.first_packet_time_ms = clock_->TimeInMilliseconds();
switch (rtcp_block.type()) {
case rtcp::SenderReport::kPacketType:
HandleSenderReport(rtcp_block, packet_information);
received_blocks[packet_information->remote_ssrc].sender_report = true;
break;
case rtcp::ReceiverReport::kPacketType:
HandleReceiverReport(rtcp_block, packet_information);
break;
case rtcp::Sdes::kPacketType:
HandleSdes(rtcp_block, packet_information);
break;
case rtcp::ExtendedReports::kPacketType: {
bool contains_dlrr = false;
uint32_t ssrc = 0;
HandleXr(rtcp_block, packet_information, contains_dlrr, ssrc);
if (contains_dlrr) {
received_blocks[ssrc].dlrr = true;
}
break;
}
case rtcp::Bye::kPacketType:
HandleBye(rtcp_block);
break;
case rtcp::App::kPacketType:
HandleApp(rtcp_block, packet_information);
break;
case rtcp::Rtpfb::kPacketType:
switch (rtcp_block.fmt()) {
case rtcp::Nack::kFeedbackMessageType:
HandleNack(rtcp_block, packet_information);
break;
case rtcp::Tmmbr::kFeedbackMessageType:
HandleTmmbr(rtcp_block, packet_information);
break;
case rtcp::Tmmbn::kFeedbackMessageType:
HandleTmmbn(rtcp_block, packet_information);
break;
case rtcp::RapidResyncRequest::kFeedbackMessageType:
HandleSrReq(rtcp_block, packet_information);
break;
case rtcp::TransportFeedback::kFeedbackMessageType:
HandleTransportFeedback(rtcp_block, packet_information);
break;
default:
++num_skipped_packets_;
break;
}
break;
case rtcp::Psfb::kPacketType:
switch (rtcp_block.fmt()) {
case rtcp::Pli::kFeedbackMessageType:
HandlePli(rtcp_block, packet_information);
break;
case rtcp::Fir::kFeedbackMessageType:
HandleFir(rtcp_block, packet_information);
break;
case rtcp::Psfb::kAfbMessageType:
HandlePsfbApp(rtcp_block, packet_information);
break;
default:
++num_skipped_packets_;
break;
}
break;
default:
++num_skipped_packets_;
break;
}
}
for (const auto& rb : received_blocks) {
if (rb.second.sender_report && !rb.second.dlrr) {
auto rtt_stats = non_sender_rtts_.find(rb.first);
if (rtt_stats != non_sender_rtts_.end()) {
rtt_stats->second.Invalidate();
}
}
}
if (packet_type_counter_observer_) {
packet_type_counter_observer_->RtcpPacketTypesCounterUpdated(
main_ssrc_, packet_type_counter_);
}
if (num_skipped_packets_ > 0) {
const int64_t now_ms = clock_->TimeInMilliseconds();
if (now_ms - last_skipped_packets_warning_ms_ >= kMaxWarningLogIntervalMs) {
last_skipped_packets_warning_ms_ = now_ms;
RTC_LOG(LS_WARNING)
<< num_skipped_packets_
<< " RTCP blocks were skipped due to being malformed or of "
"unrecognized/unsupported type, during the past "
<< (kMaxWarningLogIntervalMs / 1000) << " second period.";
}
}
return true;
}
void RTCPReceiver::HandleSenderReport(const CommonHeader& rtcp_block,
PacketInformation* packet_information) {
rtcp::SenderReport sender_report;
if (!sender_report.Parse(rtcp_block)) {
++num_skipped_packets_;
return;
}
const uint32_t remote_ssrc = sender_report.sender_ssrc();
packet_information->remote_ssrc = remote_ssrc;
UpdateTmmbrRemoteIsAlive(remote_ssrc);
// Have I received RTP packets from this party?
if (remote_ssrc_ == remote_ssrc) {
// Only signal that we have received a SR when we accept one.
packet_information->packet_type_flags |= kRtcpSr;
remote_sender_ntp_time_ = sender_report.ntp();
remote_sender_rtp_time_ = sender_report.rtp_timestamp();
last_received_sr_ntp_ = clock_->CurrentNtpTime();
remote_sender_packet_count_ = sender_report.sender_packet_count();
remote_sender_octet_count_ = sender_report.sender_octet_count();
remote_sender_reports_count_++;
} else {
// We will only store the send report from one source, but
// we will store all the receive blocks.
packet_information->packet_type_flags |= kRtcpRr;
}
for (const rtcp::ReportBlock& report_block : sender_report.report_blocks())
HandleReportBlock(report_block, packet_information, remote_ssrc);
}
void RTCPReceiver::HandleReceiverReport(const CommonHeader& rtcp_block,
PacketInformation* packet_information) {
rtcp::ReceiverReport receiver_report;
if (!receiver_report.Parse(rtcp_block)) {
++num_skipped_packets_;
return;
}
const uint32_t remote_ssrc = receiver_report.sender_ssrc();
packet_information->remote_ssrc = remote_ssrc;
UpdateTmmbrRemoteIsAlive(remote_ssrc);
packet_information->packet_type_flags |= kRtcpRr;
for (const ReportBlock& report_block : receiver_report.report_blocks())
HandleReportBlock(report_block, packet_information, remote_ssrc);
}
void RTCPReceiver::HandleReportBlock(const ReportBlock& report_block,
PacketInformation* packet_information,
uint32_t remote_ssrc) {
// This will be called once per report block in the RTCP packet.
// We filter out all report blocks that are not for us.
// Each packet has max 31 RR blocks.
//
// We can calc RTT if we send a send report and get a report block back.
// `report_block.source_ssrc()` is the SSRC identifier of the source to
// which the information in this reception report block pertains.
// Filter out all report blocks that are not for us.
if (!registered_ssrcs_.contains(report_block.source_ssrc()))
return;
last_received_rb_ = clock_->CurrentTime();
ReportBlockData* report_block_data =
&received_report_blocks_[report_block.source_ssrc()];
RTCPReportBlock rtcp_report_block;
rtcp_report_block.sender_ssrc = remote_ssrc;
rtcp_report_block.source_ssrc = report_block.source_ssrc();
rtcp_report_block.fraction_lost = report_block.fraction_lost();
rtcp_report_block.packets_lost = report_block.cumulative_lost_signed();
if (report_block.extended_high_seq_num() >
report_block_data->report_block().extended_highest_sequence_number) {
// We have successfully delivered new RTP packets to the remote side after
// the last RR was sent from the remote side.
last_increased_sequence_number_ = last_received_rb_;
}
rtcp_report_block.extended_highest_sequence_number =
report_block.extended_high_seq_num();
rtcp_report_block.jitter = report_block.jitter();
rtcp_report_block.delay_since_last_sender_report =
report_block.delay_since_last_sr();
rtcp_report_block.last_sender_report_timestamp = report_block.last_sr();
report_block_data->SetReportBlock(rtcp_report_block, rtc::TimeUTCMicros());
int64_t rtt_ms = 0;
uint32_t send_time_ntp = report_block.last_sr();
// RFC3550, section 6.4.1, LSR field discription states:
// If no SR has been received yet, the field is set to zero.
// Receiver rtp_rtcp module is not expected to calculate rtt using
// Sender Reports even if it accidentally can.
// TODO(nisse): Use this way to determine the RTT only when `receiver_only_`
// is false. However, that currently breaks the tests of the
// googCaptureStartNtpTimeMs stat for audio receive streams. To fix, either
// delete all dependencies on RTT measurements for audio receive streams, or
// ensure that audio receive streams that need RTT and stats that depend on it
// are configured with an associated audio send stream.
if (send_time_ntp != 0) {
uint32_t delay_ntp = report_block.delay_since_last_sr();
// Local NTP time.
uint32_t receive_time_ntp =
CompactNtp(clock_->ConvertTimestampToNtpTime(last_received_rb_));
// RTT in 1/(2^16) seconds.
uint32_t rtt_ntp = receive_time_ntp - delay_ntp - send_time_ntp;
// Convert to 1/1000 seconds (milliseconds).
rtt_ms = CompactNtpRttToMs(rtt_ntp);
report_block_data->AddRoundTripTimeSample(rtt_ms);
if (report_block.source_ssrc() == main_ssrc_) {
rtts_[remote_ssrc].AddRtt(TimeDelta::Millis(rtt_ms));
}
packet_information->rtt_ms = rtt_ms;
}
packet_information->report_blocks.push_back(
report_block_data->report_block());
packet_information->report_block_datas.push_back(*report_block_data);
}
RTCPReceiver::TmmbrInformation* RTCPReceiver::FindOrCreateTmmbrInfo(
uint32_t remote_ssrc) {
// Create or find receive information.
TmmbrInformation* tmmbr_info = &tmmbr_infos_[remote_ssrc];
// Update that this remote is alive.
tmmbr_info->last_time_received_ms = clock_->TimeInMilliseconds();
return tmmbr_info;
}
void RTCPReceiver::UpdateTmmbrRemoteIsAlive(uint32_t remote_ssrc) {
auto tmmbr_it = tmmbr_infos_.find(remote_ssrc);
if (tmmbr_it != tmmbr_infos_.end())
tmmbr_it->second.last_time_received_ms = clock_->TimeInMilliseconds();
}
RTCPReceiver::TmmbrInformation* RTCPReceiver::GetTmmbrInformation(
uint32_t remote_ssrc) {
auto it = tmmbr_infos_.find(remote_ssrc);
if (it == tmmbr_infos_.end())
return nullptr;
return &it->second;
}
// These two methods (RtcpRrTimeout and RtcpRrSequenceNumberTimeout) only exist
// for tests and legacy code (rtp_rtcp_impl.cc). We should be able to to delete
// the methods and require that access to the locked variables only happens on
// the worker thread and thus no locking is needed.
bool RTCPReceiver::RtcpRrTimeout() {
MutexLock lock(&rtcp_receiver_lock_);
return RtcpRrTimeoutLocked(clock_->CurrentTime());
}
bool RTCPReceiver::RtcpRrSequenceNumberTimeout() {
MutexLock lock(&rtcp_receiver_lock_);
return RtcpRrSequenceNumberTimeoutLocked(clock_->CurrentTime());
}
bool RTCPReceiver::UpdateTmmbrTimers() {
MutexLock lock(&rtcp_receiver_lock_);
int64_t now_ms = clock_->TimeInMilliseconds();
int64_t timeout_ms = now_ms - kTmmbrTimeoutIntervalMs;
if (oldest_tmmbr_info_ms_ >= timeout_ms)
return false;
bool update_bounding_set = false;
oldest_tmmbr_info_ms_ = -1;
for (auto tmmbr_it = tmmbr_infos_.begin(); tmmbr_it != tmmbr_infos_.end();) {
TmmbrInformation* tmmbr_info = &tmmbr_it->second;
if (tmmbr_info->last_time_received_ms > 0) {
if (tmmbr_info->last_time_received_ms < timeout_ms) {
// No rtcp packet for the last 5 regular intervals, reset limitations.
tmmbr_info->tmmbr.clear();
// Prevent that we call this over and over again.
tmmbr_info->last_time_received_ms = 0;
// Send new TMMBN to all channels using the default codec.
update_bounding_set = true;
} else if (oldest_tmmbr_info_ms_ == -1 ||
tmmbr_info->last_time_received_ms < oldest_tmmbr_info_ms_) {
oldest_tmmbr_info_ms_ = tmmbr_info->last_time_received_ms;
}
++tmmbr_it;
} else if (tmmbr_info->ready_for_delete) {
// When we dont have a last_time_received_ms and the object is marked
// ready_for_delete it's removed from the map.
tmmbr_it = tmmbr_infos_.erase(tmmbr_it);
} else {
++tmmbr_it;
}
}
return update_bounding_set;
}
std::vector<rtcp::TmmbItem> RTCPReceiver::BoundingSet(bool* tmmbr_owner) {
MutexLock lock(&rtcp_receiver_lock_);
TmmbrInformation* tmmbr_info = GetTmmbrInformation(remote_ssrc_);
if (!tmmbr_info)
return std::vector<rtcp::TmmbItem>();
*tmmbr_owner = TMMBRHelp::IsOwner(tmmbr_info->tmmbn, main_ssrc_);
return tmmbr_info->tmmbn;
}
void RTCPReceiver::HandleSdes(const CommonHeader& rtcp_block,
PacketInformation* packet_information) {
rtcp::Sdes sdes;
if (!sdes.Parse(rtcp_block)) {
++num_skipped_packets_;
return;
}
for (const rtcp::Sdes::Chunk& chunk : sdes.chunks()) {
if (cname_callback_)
cname_callback_->OnCname(chunk.ssrc, chunk.cname);
}
packet_information->packet_type_flags |= kRtcpSdes;
}
void RTCPReceiver::HandleNack(const CommonHeader& rtcp_block,
PacketInformation* packet_information) {
rtcp::Nack nack;
if (!nack.Parse(rtcp_block)) {
++num_skipped_packets_;
return;
}
if (receiver_only_ || main_ssrc_ != nack.media_ssrc()) // Not to us.
return;
packet_information->nack_sequence_numbers.insert(
packet_information->nack_sequence_numbers.end(),
nack.packet_ids().begin(), nack.packet_ids().end());
for (uint16_t packet_id : nack.packet_ids())
nack_stats_.ReportRequest(packet_id);
if (!nack.packet_ids().empty()) {
packet_information->packet_type_flags |= kRtcpNack;
++packet_type_counter_.nack_packets;
packet_type_counter_.nack_requests = nack_stats_.requests();
packet_type_counter_.unique_nack_requests = nack_stats_.unique_requests();
}
}
void RTCPReceiver::HandleApp(const rtcp::CommonHeader& rtcp_block,
PacketInformation* packet_information) {
rtcp::App app;
if (app.Parse(rtcp_block)) {
if (app.name() == rtcp::RemoteEstimate::kName &&
app.sub_type() == rtcp::RemoteEstimate::kSubType) {
rtcp::RemoteEstimate estimate(std::move(app));
if (estimate.ParseData()) {
packet_information->network_state_estimate = estimate.estimate();
return;
}
}
}
++num_skipped_packets_;
}
void RTCPReceiver::HandleBye(const CommonHeader& rtcp_block) {
rtcp::Bye bye;
if (!bye.Parse(rtcp_block)) {
++num_skipped_packets_;
return;
}
// Clear our lists.
rtts_.erase(bye.sender_ssrc());
EraseIf(received_report_blocks_, [&](const auto& elem) {
return elem.second.report_block().sender_ssrc == bye.sender_ssrc();
});
TmmbrInformation* tmmbr_info = GetTmmbrInformation(bye.sender_ssrc());
if (tmmbr_info)
tmmbr_info->ready_for_delete = true;
last_fir_.erase(bye.sender_ssrc());
auto it = received_rrtrs_ssrc_it_.find(bye.sender_ssrc());
if (it != received_rrtrs_ssrc_it_.end()) {
received_rrtrs_.erase(it->second);
received_rrtrs_ssrc_it_.erase(it);
}
xr_rr_rtt_ms_ = 0;
}
void RTCPReceiver::HandleXr(const CommonHeader& rtcp_block,
PacketInformation* packet_information,
bool& contains_dlrr,
uint32_t& ssrc) {
rtcp::ExtendedReports xr;
if (!xr.Parse(rtcp_block)) {
++num_skipped_packets_;
return;
}
ssrc = xr.sender_ssrc();
contains_dlrr = !xr.dlrr().sub_blocks().empty();
if (xr.rrtr())
HandleXrReceiveReferenceTime(xr.sender_ssrc(), *xr.rrtr());
for (const rtcp::ReceiveTimeInfo& time_info : xr.dlrr().sub_blocks())
HandleXrDlrrReportBlock(xr.sender_ssrc(), time_info);
if (xr.target_bitrate()) {
HandleXrTargetBitrate(xr.sender_ssrc(), *xr.target_bitrate(),
packet_information);
}
}
void RTCPReceiver::HandleXrReceiveReferenceTime(uint32_t sender_ssrc,
const rtcp::Rrtr& rrtr) {
uint32_t received_remote_mid_ntp_time = CompactNtp(rrtr.ntp());
uint32_t local_receive_mid_ntp_time = CompactNtp(clock_->CurrentNtpTime());
auto it = received_rrtrs_ssrc_it_.find(sender_ssrc);
if (it != received_rrtrs_ssrc_it_.end()) {
it->second->received_remote_mid_ntp_time = received_remote_mid_ntp_time;
it->second->local_receive_mid_ntp_time = local_receive_mid_ntp_time;
} else {
if (received_rrtrs_.size() < kMaxNumberOfStoredRrtrs) {
received_rrtrs_.emplace_back(sender_ssrc, received_remote_mid_ntp_time,
local_receive_mid_ntp_time);
received_rrtrs_ssrc_it_[sender_ssrc] = std::prev(received_rrtrs_.end());
} else {
RTC_LOG(LS_WARNING) << "Discarding received RRTR for ssrc " << sender_ssrc
<< ", reached maximum number of stored RRTRs.";
}
}
}
void RTCPReceiver::HandleXrDlrrReportBlock(uint32_t sender_ssrc,
const rtcp::ReceiveTimeInfo& rti) {
if (!registered_ssrcs_.contains(rti.ssrc)) // Not to us.
return;
// Caller should explicitly enable rtt calculation using extended reports.
if (!xr_rrtr_status_)
return;
// The send_time and delay_rr fields are in units of 1/2^16 sec.
uint32_t send_time_ntp = rti.last_rr;
// RFC3611, section 4.5, LRR field discription states:
// If no such block has been received, the field is set to zero.
if (send_time_ntp == 0) {
auto rtt_stats = non_sender_rtts_.find(sender_ssrc);
if (rtt_stats != non_sender_rtts_.end()) {
rtt_stats->second.Invalidate();
}
return;
}
uint32_t delay_ntp = rti.delay_since_last_rr;
uint32_t now_ntp = CompactNtp(clock_->CurrentNtpTime());
uint32_t rtt_ntp = now_ntp - delay_ntp - send_time_ntp;
xr_rr_rtt_ms_ = CompactNtpRttToMs(rtt_ntp);
non_sender_rtts_[sender_ssrc].Update(TimeDelta::Millis(xr_rr_rtt_ms_));
}
void RTCPReceiver::HandleXrTargetBitrate(
uint32_t ssrc,
const rtcp::TargetBitrate& target_bitrate,
PacketInformation* packet_information) {
if (ssrc != remote_ssrc_) {
return; // Not for us.
}
VideoBitrateAllocation bitrate_allocation;
for (const auto& item : target_bitrate.GetTargetBitrates()) {
if (item.spatial_layer >= kMaxSpatialLayers ||
item.temporal_layer >= kMaxTemporalStreams) {
RTC_LOG(LS_WARNING)
<< "Invalid layer in XR target bitrate pack: spatial index "
<< item.spatial_layer << ", temporal index " << item.temporal_layer
<< ", dropping.";
} else {
bitrate_allocation.SetBitrate(item.spatial_layer, item.temporal_layer,
item.target_bitrate_kbps * 1000);
}
}
packet_information->target_bitrate_allocation.emplace(bitrate_allocation);
}
void RTCPReceiver::HandlePli(const CommonHeader& rtcp_block,
PacketInformation* packet_information) {
rtcp::Pli pli;
if (!pli.Parse(rtcp_block)) {
++num_skipped_packets_;
return;
}
if (main_ssrc_ == pli.media_ssrc()) {
++packet_type_counter_.pli_packets;
// Received a signal that we need to send a new key frame.
packet_information->packet_type_flags |= kRtcpPli;
}
}
void RTCPReceiver::HandleTmmbr(const CommonHeader& rtcp_block,
PacketInformation* packet_information) {
rtcp::Tmmbr tmmbr;
if (!tmmbr.Parse(rtcp_block)) {
++num_skipped_packets_;
return;
}
uint32_t sender_ssrc = tmmbr.sender_ssrc();
if (tmmbr.media_ssrc()) {
// media_ssrc() SHOULD be 0 if same as SenderSSRC.
// In relay mode this is a valid number.
sender_ssrc = tmmbr.media_ssrc();
}
for (const rtcp::TmmbItem& request : tmmbr.requests()) {
if (main_ssrc_ != request.ssrc() || request.bitrate_bps() == 0)
continue;
TmmbrInformation* tmmbr_info = FindOrCreateTmmbrInfo(tmmbr.sender_ssrc());
auto* entry = &tmmbr_info->tmmbr[sender_ssrc];
entry->tmmbr_item = rtcp::TmmbItem(sender_ssrc, request.bitrate_bps(),
request.packet_overhead());
// FindOrCreateTmmbrInfo always sets `last_time_received_ms` to
// `clock_->TimeInMilliseconds()`.
entry->last_updated_ms = tmmbr_info->last_time_received_ms;
packet_information->packet_type_flags |= kRtcpTmmbr;
break;
}
}
void RTCPReceiver::HandleTmmbn(const CommonHeader& rtcp_block,
PacketInformation* packet_information) {
rtcp::Tmmbn tmmbn;
if (!tmmbn.Parse(rtcp_block)) {
++num_skipped_packets_;
return;
}
TmmbrInformation* tmmbr_info = FindOrCreateTmmbrInfo(tmmbn.sender_ssrc());
packet_information->packet_type_flags |= kRtcpTmmbn;
tmmbr_info->tmmbn = tmmbn.items();
}
void RTCPReceiver::HandleSrReq(const CommonHeader& rtcp_block,
PacketInformation* packet_information) {
rtcp::RapidResyncRequest sr_req;
if (!sr_req.Parse(rtcp_block)) {
++num_skipped_packets_;
return;
}
packet_information->packet_type_flags |= kRtcpSrReq;
}
void RTCPReceiver::HandlePsfbApp(const CommonHeader& rtcp_block,
PacketInformation* packet_information) {
{
rtcp::Remb remb;
if (remb.Parse(rtcp_block)) {
packet_information->packet_type_flags |= kRtcpRemb;
packet_information->receiver_estimated_max_bitrate_bps =
remb.bitrate_bps();
return;
}
}
{
auto loss_notification = std::make_unique<rtcp::LossNotification>();
if (loss_notification->Parse(rtcp_block)) {
packet_information->packet_type_flags |= kRtcpLossNotification;
packet_information->loss_notification = std::move(loss_notification);
return;
}
}
RTC_LOG(LS_WARNING) << "Unknown PSFB-APP packet.";
++num_skipped_packets_;
}
void RTCPReceiver::HandleFir(const CommonHeader& rtcp_block,
PacketInformation* packet_information) {
rtcp::Fir fir;
if (!fir.Parse(rtcp_block)) {
++num_skipped_packets_;
return;
}
if (fir.requests().empty())
return;
const int64_t now_ms = clock_->TimeInMilliseconds();
for (const rtcp::Fir::Request& fir_request : fir.requests()) {
// Is it our sender that is requested to generate a new keyframe.
if (main_ssrc_ != fir_request.ssrc)
continue;
++packet_type_counter_.fir_packets;
auto inserted = last_fir_.insert(std::make_pair(
fir.sender_ssrc(), LastFirStatus(now_ms, fir_request.seq_nr)));
if (!inserted.second) { // There was already an entry.
LastFirStatus* last_fir = &inserted.first->second;
// Check if we have reported this FIRSequenceNumber before.
if (fir_request.seq_nr == last_fir->sequence_number)
continue;
// Sanity: don't go crazy with the callbacks.
if (now_ms - last_fir->request_ms < kRtcpMinFrameLengthMs)
continue;
last_fir->request_ms = now_ms;
last_fir->sequence_number = fir_request.seq_nr;
}
// Received signal that we need to send a new key frame.
packet_information->packet_type_flags |= kRtcpFir;
}
}
void RTCPReceiver::HandleTransportFeedback(
const CommonHeader& rtcp_block,
PacketInformation* packet_information) {
std::unique_ptr<rtcp::TransportFeedback> transport_feedback(
new rtcp::TransportFeedback());
if (!transport_feedback->Parse(rtcp_block)) {
++num_skipped_packets_;
return;
}
packet_information->packet_type_flags |= kRtcpTransportFeedback;
packet_information->transport_feedback = std::move(transport_feedback);
}
void RTCPReceiver::NotifyTmmbrUpdated() {
// Find bounding set.
std::vector<rtcp::TmmbItem> bounding =
TMMBRHelp::FindBoundingSet(TmmbrReceived());
if (!bounding.empty() && rtcp_bandwidth_observer_) {
// We have a new bandwidth estimate on this channel.
uint64_t bitrate_bps = TMMBRHelp::CalcMinBitrateBps(bounding);
if (bitrate_bps <= std::numeric_limits<uint32_t>::max())
rtcp_bandwidth_observer_->OnReceivedEstimatedBitrate(bitrate_bps);
}
// Send tmmbn to inform remote clients about the new bandwidth.
rtp_rtcp_->SetTmmbn(std::move(bounding));
}
// Holding no Critical section.
void RTCPReceiver::TriggerCallbacksFromRtcpPacket(
const PacketInformation& packet_information) {
// Process TMMBR and REMB first to avoid multiple callbacks
// to OnNetworkChanged.
if (packet_information.packet_type_flags & kRtcpTmmbr) {
// Might trigger a OnReceivedBandwidthEstimateUpdate.
NotifyTmmbrUpdated();
}
if (!receiver_only_ && (packet_information.packet_type_flags & kRtcpSrReq)) {
rtp_rtcp_->OnRequestSendReport();
}
if (!receiver_only_ && (packet_information.packet_type_flags & kRtcpNack)) {
if (!packet_information.nack_sequence_numbers.empty()) {
RTC_LOG(LS_VERBOSE) << "Incoming NACK length: "
<< packet_information.nack_sequence_numbers.size();
rtp_rtcp_->OnReceivedNack(packet_information.nack_sequence_numbers);
}
}
// We need feedback that we have received a report block(s) so that we
// can generate a new packet in a conference relay scenario, one received
// report can generate several RTCP packets, based on number relayed/mixed
// a send report block should go out to all receivers.
if (rtcp_intra_frame_observer_) {
RTC_DCHECK(!receiver_only_);
if ((packet_information.packet_type_flags & kRtcpPli) ||
(packet_information.packet_type_flags & kRtcpFir)) {
if (packet_information.packet_type_flags & kRtcpPli) {
RTC_LOG(LS_VERBOSE)
<< "Incoming PLI from SSRC " << packet_information.remote_ssrc;
} else {
RTC_LOG(LS_VERBOSE)
<< "Incoming FIR from SSRC " << packet_information.remote_ssrc;
}
rtcp_intra_frame_observer_->OnReceivedIntraFrameRequest(main_ssrc_);
}
}
if (rtcp_loss_notification_observer_ &&
(packet_information.packet_type_flags & kRtcpLossNotification)) {
rtcp::LossNotification* loss_notification =
packet_information.loss_notification.get();
RTC_DCHECK(loss_notification);
if (loss_notification->media_ssrc() == main_ssrc_) {
rtcp_loss_notification_observer_->OnReceivedLossNotification(
loss_notification->media_ssrc(), loss_notification->last_decoded(),
loss_notification->last_received(),
loss_notification->decodability_flag());
}
}
if (rtcp_bandwidth_observer_) {
RTC_DCHECK(!receiver_only_);
if (packet_information.packet_type_flags & kRtcpRemb) {
RTC_LOG(LS_VERBOSE)
<< "Incoming REMB: "
<< packet_information.receiver_estimated_max_bitrate_bps;
rtcp_bandwidth_observer_->OnReceivedEstimatedBitrate(
packet_information.receiver_estimated_max_bitrate_bps);
}
if ((packet_information.packet_type_flags & kRtcpSr) ||
(packet_information.packet_type_flags & kRtcpRr)) {
int64_t now_ms = clock_->TimeInMilliseconds();
rtcp_bandwidth_observer_->OnReceivedRtcpReceiverReport(
packet_information.report_blocks, packet_information.rtt_ms, now_ms);
}
}
if ((packet_information.packet_type_flags & kRtcpSr) ||
(packet_information.packet_type_flags & kRtcpRr)) {
rtp_rtcp_->OnReceivedRtcpReportBlocks(packet_information.report_blocks);
}
if (transport_feedback_observer_ &&
(packet_information.packet_type_flags & kRtcpTransportFeedback)) {
uint32_t media_source_ssrc =
packet_information.transport_feedback->media_ssrc();
if (media_source_ssrc == main_ssrc_ ||
registered_ssrcs_.contains(media_source_ssrc)) {
transport_feedback_observer_->OnTransportFeedback(
*packet_information.transport_feedback);
}
}
if (network_state_estimate_observer_ &&
packet_information.network_state_estimate) {
network_state_estimate_observer_->OnRemoteNetworkEstimate(
*packet_information.network_state_estimate);
}
if (bitrate_allocation_observer_ &&
packet_information.target_bitrate_allocation) {
bitrate_allocation_observer_->OnBitrateAllocationUpdated(
*packet_information.target_bitrate_allocation);
}
if (!receiver_only_) {
if (report_block_data_observer_) {
for (const auto& report_block_data :
packet_information.report_block_datas) {
report_block_data_observer_->OnReportBlockDataUpdated(
report_block_data);
}
}
}
}
std::vector<rtcp::TmmbItem> RTCPReceiver::TmmbrReceived() {
MutexLock lock(&rtcp_receiver_lock_);
std::vector<rtcp::TmmbItem> candidates;
int64_t now_ms = clock_->TimeInMilliseconds();
int64_t timeout_ms = now_ms - kTmmbrTimeoutIntervalMs;
for (auto& kv : tmmbr_infos_) {
for (auto it = kv.second.tmmbr.begin(); it != kv.second.tmmbr.end();) {
if (it->second.last_updated_ms < timeout_ms) {
// Erase timeout entries.
it = kv.second.tmmbr.erase(it);
} else {
candidates.push_back(it->second.tmmbr_item);
++it;
}
}
}
return candidates;
}
bool RTCPReceiver::RtcpRrTimeoutLocked(Timestamp now) {
return ResetTimestampIfExpired(now, last_received_rb_, report_interval_);
}
bool RTCPReceiver::RtcpRrSequenceNumberTimeoutLocked(Timestamp now) {
return ResetTimestampIfExpired(now, last_increased_sequence_number_,
report_interval_);
}
} // namespace webrtc