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webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
Nico Grunbaum 7eac6caeee Don't use wall clock for stats 
This uses the local NTP clock for RTCP report block stats.

This code exists in the version that Mozilla is shipping, with a review
here https://phabricator.services.mozilla.com/D127709 .

Bug: webrtc:13484
Change-Id: I2f46ec02acab0bbb09040778b05b248c2d815bd1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/240142
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35787}
2022-01-25 15:39:53 +00:00

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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.
*/
#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();
// Number of seconds since 1900 January 1 00:00 GMT (see
// https://tools.ietf.org/html/rfc868).
report_block_data->SetReportBlock(
rtcp_report_block,
(clock_->CurrentNtpInMilliseconds() - rtc::kNtpJan1970Millisecs) *
rtc::kNumMicrosecsPerMillisec);
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
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