webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc

/*
 *  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 "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"

#include <string.h>

#include <limits>
#include <map>
#include <memory>
#include <utility>
#include <vector>

#include "webrtc/common_types.h"
#include "webrtc/common_video/include/video_bitrate_allocator.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/fir.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "webrtc/modules/rtp_rtcp/source/time_util.h"
#include "webrtc/modules/rtp_rtcp/source/tmmbr_help.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/logging.h"
#include "webrtc/rtc_base/trace_event.h"
#include "webrtc/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 kMaxWarningLogIntervalMs = 10000;
const int64_t kRtcpMinFrameLengthMs = 17;

}  // namespace

struct RTCPReceiver::PacketInformation {
  uint32_t packet_type_flags = 0;  // RTCPPacketTypeFlags bit field.

  uint32_t remote_ssrc = 0;
  std::vector<uint16_t> nack_sequence_numbers;
  ReportBlockList report_blocks;
  int64_t rtt_ms = 0;
  uint32_t receiver_estimated_max_bitrate_bps = 0;
  std::unique_ptr<rtcp::TransportFeedback> transport_feedback;
  rtc::Optional<BitrateAllocation> target_bitrate_allocation;
};

// Structure for handing TMMBR and TMMBN rtcp messages (RFC5104, section 3.5.4).
struct RTCPReceiver::TmmbrInformation {
  struct TimedTmmbrItem {
    rtcp::TmmbItem tmmbr_item;
    int64_t last_updated_ms;
  };

  int64_t last_time_received_ms = 0;

  bool ready_for_delete = false;

  std::vector<rtcp::TmmbItem> tmmbn;
  std::map<uint32_t, TimedTmmbrItem> tmmbr;
};

struct RTCPReceiver::ReportBlockWithRtt {
  RTCPReportBlock report_block;

  int64_t last_rtt_ms = 0;
  int64_t min_rtt_ms = 0;
  int64_t max_rtt_ms = 0;
  int64_t sum_rtt_ms = 0;
  size_t num_rtts = 0;
};

struct RTCPReceiver::LastFirStatus {
  LastFirStatus(int64_t now_ms, uint8_t sequence_number)
      : request_ms(now_ms), sequence_number(sequence_number) {}
  int64_t request_ms;
  uint8_t sequence_number;
};

RTCPReceiver::RTCPReceiver(
    Clock* clock,
    bool receiver_only,
    RtcpPacketTypeCounterObserver* packet_type_counter_observer,
    RtcpBandwidthObserver* rtcp_bandwidth_observer,
    RtcpIntraFrameObserver* rtcp_intra_frame_observer,
    TransportFeedbackObserver* transport_feedback_observer,
    VideoBitrateAllocationObserver* bitrate_allocation_observer,
    ModuleRtpRtcp* owner)
    : clock_(clock),
      receiver_only_(receiver_only),
      rtp_rtcp_(owner),
      rtcp_bandwidth_observer_(rtcp_bandwidth_observer),
      rtcp_intra_frame_observer_(rtcp_intra_frame_observer),
      transport_feedback_observer_(transport_feedback_observer),
      bitrate_allocation_observer_(bitrate_allocation_observer),
      main_ssrc_(0),
      remote_ssrc_(0),
      xr_rrtr_status_(false),
      xr_rr_rtt_ms_(0),
      oldest_tmmbr_info_ms_(0),
      last_received_rr_ms_(0),
      last_increased_sequence_number_ms_(0),
      stats_callback_(nullptr),
      packet_type_counter_observer_(packet_type_counter_observer),
      num_skipped_packets_(0),
      last_skipped_packets_warning_ms_(clock->TimeInMilliseconds()) {
  RTC_DCHECK(owner);
  memset(&remote_sender_info_, 0, sizeof(remote_sender_info_));
}

RTCPReceiver::~RTCPReceiver() {}

bool RTCPReceiver::IncomingPacket(const uint8_t* packet, size_t packet_size) {
  if (packet_size == 0) {
    LOG(LS_WARNING) << "Incoming empty RTCP packet";
    return false;
  }

  PacketInformation packet_information;
  if (!ParseCompoundPacket(packet, packet + packet_size, &packet_information))
    return false;
  TriggerCallbacksFromRtcpPacket(packet_information);
  return true;
}

int64_t RTCPReceiver::LastReceivedReceiverReport() const {
  rtc::CritScope lock(&rtcp_receiver_lock_);
  return last_received_rr_ms_;
}

void RTCPReceiver::SetRemoteSSRC(uint32_t ssrc) {
  rtc::CritScope lock(&rtcp_receiver_lock_);
  // New SSRC reset old reports.
  memset(&remote_sender_info_, 0, sizeof(remote_sender_info_));
  last_received_sr_ntp_.Reset();
  remote_ssrc_ = ssrc;
}

uint32_t RTCPReceiver::RemoteSSRC() const {
  rtc::CritScope lock(&rtcp_receiver_lock_);
  return remote_ssrc_;
}

void RTCPReceiver::SetSsrcs(uint32_t main_ssrc,
                            const std::set<uint32_t>& registered_ssrcs) {
  rtc::CritScope lock(&rtcp_receiver_lock_);
  main_ssrc_ = main_ssrc;
  registered_ssrcs_ = registered_ssrcs;
}

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 {
  rtc::CritScope lock(&rtcp_receiver_lock_);

  auto it = received_report_blocks_.find(main_ssrc_);
  if (it == received_report_blocks_.end())
    return -1;

  auto it_info = it->second.find(remote_ssrc);
  if (it_info == it->second.end())
    return -1;

  const ReportBlockWithRtt* report_block = &it_info->second;

  if (report_block->num_rtts == 0)
    return -1;

  if (last_rtt_ms)
    *last_rtt_ms = report_block->last_rtt_ms;

  if (avg_rtt_ms)
    *avg_rtt_ms = report_block->sum_rtt_ms / report_block->num_rtts;

  if (min_rtt_ms)
    *min_rtt_ms = report_block->min_rtt_ms;

  if (max_rtt_ms)
    *max_rtt_ms = report_block->max_rtt_ms;

  return 0;
}

void RTCPReceiver::SetRtcpXrRrtrStatus(bool enable) {
  rtc::CritScope lock(&rtcp_receiver_lock_);
  xr_rrtr_status_ = enable;
}

bool RTCPReceiver::GetAndResetXrRrRtt(int64_t* rtt_ms) {
  RTC_DCHECK(rtt_ms);
  rtc::CritScope 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;
}

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) const {
  rtc::CritScope 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_info_.NTPseconds;
  if (received_ntp_frac)
    *received_ntp_frac = remote_sender_info_.NTPfraction;

  // Rtp time from incoming SenderReport.
  if (rtcp_timestamp)
    *rtcp_timestamp = remote_sender_info_.RTPtimeStamp;

  // 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();

  return true;
}

bool RTCPReceiver::LastReceivedXrReferenceTimeInfo(
    rtcp::ReceiveTimeInfo* info) const {
  RTC_DCHECK(info);
  rtc::CritScope lock(&rtcp_receiver_lock_);
  if (!last_received_xr_ntp_.Valid())
    return false;

  info->ssrc = remote_time_info_.ssrc;
  info->last_rr = remote_time_info_.last_rr;

  // Get the delay since last received report (RFC 3611).
  uint32_t receive_time_ntp = CompactNtp(last_received_xr_ntp_);
  uint32_t now_ntp = CompactNtp(clock_->CurrentNtpTime());

  info->delay_since_last_rr = now_ntp - receive_time_ntp;
  return true;
}

int32_t RTCPReceiver::SenderInfoReceived(RTCPSenderInfo* sender_info) const {
  RTC_DCHECK(sender_info);
  rtc::CritScope lock(&rtcp_receiver_lock_);
  if (!last_received_sr_ntp_.Valid())
    return -1;

  memcpy(sender_info, &remote_sender_info_, sizeof(RTCPSenderInfo));
  return 0;
}

// We can get multiple receive reports when we receive the report from a CE.
int32_t RTCPReceiver::StatisticsReceived(
    std::vector<RTCPReportBlock>* receive_blocks) const {
  RTC_DCHECK(receive_blocks);
  rtc::CritScope lock(&rtcp_receiver_lock_);
  for (const auto& reports_per_receiver : received_report_blocks_)
    for (const auto& report : reports_per_receiver.second)
      receive_blocks->push_back(report.second.report_block);
  return 0;
}

bool RTCPReceiver::ParseCompoundPacket(const uint8_t* packet_begin,
                                       const uint8_t* packet_end,
                                       PacketInformation* packet_information) {
  rtc::CritScope lock(&rtcp_receiver_lock_);

  CommonHeader rtcp_block;
  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.
        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);
        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:
        HandleXr(rtcp_block, packet_information);
        break;
      case rtcp::Bye::kPacketType:
        HandleBye(rtcp_block);
        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::Remb::kFeedbackMessageType:
            HandlePsfbApp(rtcp_block, packet_information);
            break;
          default:
            ++num_skipped_packets_;
            break;
        }
        break;
      default:
        ++num_skipped_packets_;
        break;
    }
  }

  if (packet_type_counter_observer_) {
    packet_type_counter_observer_->RtcpPacketTypesCounterUpdated(
        main_ssrc_, packet_type_counter_);
  }

  int64_t now_ms = clock_->TimeInMilliseconds();
  if (now_ms - last_skipped_packets_warning_ms_ >= kMaxWarningLogIntervalMs &&
      num_skipped_packets_ > 0) {
    last_skipped_packets_warning_ms_ = now_ms;
    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);

  TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "SR",
                       "remote_ssrc", remote_ssrc, "ssrc", main_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;

    // Save the NTP time of this report.
    remote_sender_info_.NTPseconds = sender_report.ntp().seconds();
    remote_sender_info_.NTPfraction = sender_report.ntp().fractions();
    remote_sender_info_.RTPtimeStamp = sender_report.rtp_timestamp();
    remote_sender_info_.sendPacketCount = sender_report.sender_packet_count();
    remote_sender_info_.sendOctetCount = sender_report.sender_octet_count();

    last_received_sr_ntp_ = clock_->CurrentNtpTime();
  } 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;
  }

  last_received_rr_ms_ = clock_->TimeInMilliseconds();

  const uint32_t remote_ssrc = receiver_report.sender_ssrc();

  packet_information->remote_ssrc = remote_ssrc;

  UpdateTmmbrRemoteIsAlive(remote_ssrc);

  TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "RR",
                       "remote_ssrc", remote_ssrc, "ssrc", main_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_.count(report_block.source_ssrc()) == 0)
    return;

  ReportBlockWithRtt* report_block_info =
      &received_report_blocks_[report_block.source_ssrc()][remote_ssrc];
  report_block_info->report_block.remoteSSRC = remote_ssrc;
  report_block_info->report_block.sourceSSRC = report_block.source_ssrc();
  report_block_info->report_block.fractionLost = report_block.fraction_lost();
  report_block_info->report_block.cumulativeLost =
      report_block.cumulative_lost();
  if (report_block.extended_high_seq_num() >
      report_block_info->report_block.extendedHighSeqNum) {
    // 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_ms_ = clock_->TimeInMilliseconds();
  }
  report_block_info->report_block.extendedHighSeqNum =
      report_block.extended_high_seq_num();
  report_block_info->report_block.jitter = report_block.jitter();
  report_block_info->report_block.delaySinceLastSR =
      report_block.delay_since_last_sr();
  report_block_info->report_block.lastSR = report_block.last_sr();

  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.
  if (!receiver_only_ && send_time_ntp != 0) {
    uint32_t delay_ntp = report_block.delay_since_last_sr();
    // Local NTP time.
    uint32_t receive_time_ntp = CompactNtp(clock_->CurrentNtpTime());

    // 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);
    if (rtt_ms > report_block_info->max_rtt_ms)
      report_block_info->max_rtt_ms = rtt_ms;

    if (report_block_info->num_rtts == 0 ||
        rtt_ms < report_block_info->min_rtt_ms)
      report_block_info->min_rtt_ms = rtt_ms;

    report_block_info->last_rtt_ms = rtt_ms;
    report_block_info->sum_rtt_ms += rtt_ms;
    ++report_block_info->num_rtts;
  }

  TRACE_COUNTER_ID1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "RR_RTT",
                    report_block.source_ssrc(), rtt_ms);

  packet_information->rtt_ms = rtt_ms;
  packet_information->report_blocks.push_back(report_block_info->report_block);
}

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;
}

bool RTCPReceiver::RtcpRrTimeout(int64_t rtcp_interval_ms) {
  rtc::CritScope lock(&rtcp_receiver_lock_);
  if (last_received_rr_ms_ == 0)
    return false;

  int64_t time_out_ms = kRrTimeoutIntervals * rtcp_interval_ms;
  if (clock_->TimeInMilliseconds() > last_received_rr_ms_ + time_out_ms) {
    // Reset the timer to only trigger one log.
    last_received_rr_ms_ = 0;
    return true;
  }
  return false;
}

bool RTCPReceiver::RtcpRrSequenceNumberTimeout(int64_t rtcp_interval_ms) {
  rtc::CritScope lock(&rtcp_receiver_lock_);
  if (last_increased_sequence_number_ms_ == 0)
    return false;

  int64_t time_out_ms = kRrTimeoutIntervals * rtcp_interval_ms;
  if (clock_->TimeInMilliseconds() >
      last_increased_sequence_number_ms_ + time_out_ms) {
    // Reset the timer to only trigger one log.
    last_increased_sequence_number_ms_ = 0;
    return true;
  }
  return false;
}

bool RTCPReceiver::UpdateTmmbrTimers() {
  rtc::CritScope lock(&rtcp_receiver_lock_);

  int64_t now_ms = clock_->TimeInMilliseconds();
  // Use audio define since we don't know what interval the remote peer use.
  int64_t timeout_ms = now_ms - 5 * RTCP_INTERVAL_AUDIO_MS;

  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) {
  rtc::CritScope 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()) {
    received_cnames_[chunk.ssrc] = chunk.cname;
    {
      rtc::CritScope lock(&feedbacks_lock_);
      if (stats_callback_)
        stats_callback_->CNameChanged(chunk.cname.c_str(), chunk.ssrc);
    }
  }
  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::HandleBye(const CommonHeader& rtcp_block) {
  rtcp::Bye bye;
  if (!bye.Parse(rtcp_block)) {
    ++num_skipped_packets_;
    return;
  }

  // Clear our lists.
  for (auto& reports_per_receiver : received_report_blocks_)
    reports_per_receiver.second.erase(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());
  received_cnames_.erase(bye.sender_ssrc());
  xr_rr_rtt_ms_ = 0;
}

void RTCPReceiver::HandleXr(const CommonHeader& rtcp_block,
                            PacketInformation* packet_information) {
  rtcp::ExtendedReports xr;
  if (!xr.Parse(rtcp_block)) {
    ++num_skipped_packets_;
    return;
  }

  if (xr.rrtr())
    HandleXrReceiveReferenceTime(xr.sender_ssrc(), *xr.rrtr());

  for (const rtcp::ReceiveTimeInfo& time_info : xr.dlrr().sub_blocks())
    HandleXrDlrrReportBlock(time_info);

  if (xr.target_bitrate())
    HandleXrTargetBitrate(*xr.target_bitrate(), packet_information);
}

void RTCPReceiver::HandleXrReceiveReferenceTime(uint32_t sender_ssrc,
                                                const rtcp::Rrtr& rrtr) {
  remote_time_info_.ssrc = sender_ssrc;
  remote_time_info_.last_rr = CompactNtp(rrtr.ntp());
  last_received_xr_ntp_ = clock_->CurrentNtpTime();
}

void RTCPReceiver::HandleXrDlrrReportBlock(const rtcp::ReceiveTimeInfo& rti) {
  if (registered_ssrcs_.count(rti.ssrc) == 0)  // 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)
    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);
}

void RTCPReceiver::HandleXrTargetBitrate(
    const rtcp::TargetBitrate& target_bitrate,
    PacketInformation* packet_information) {
  BitrateAllocation bitrate_allocation;
  for (const auto& item : target_bitrate.GetTargetBitrates()) {
    if (item.spatial_layer >= kMaxSpatialLayers ||
        item.temporal_layer >= kMaxTemporalStreams) {
      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()) {
    TRACE_EVENT_INSTANT0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "PLI");

    ++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());
    entry->last_updated_ms = clock_->TimeInMilliseconds();

    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;
  }

  ++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;
  }

  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;

    int64_t now_ms = clock_->TimeInMilliseconds();
    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));
}

void RTCPReceiver::RegisterRtcpStatisticsCallback(
    RtcpStatisticsCallback* callback) {
  rtc::CritScope cs(&feedbacks_lock_);
  stats_callback_ = callback;
}

RtcpStatisticsCallback* RTCPReceiver::GetRtcpStatisticsCallback() {
  rtc::CritScope cs(&feedbacks_lock_);
  return stats_callback_;
}

// 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();
  }
  uint32_t local_ssrc;
  std::set<uint32_t> registered_ssrcs;
  {
    // We don't want to hold this critsect when triggering the callbacks below.
    rtc::CritScope lock(&rtcp_receiver_lock_);
    local_ssrc = main_ssrc_;
    registered_ssrcs = registered_ssrcs_;
  }
  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()) {
      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) {
        LOG(LS_VERBOSE) << "Incoming PLI from SSRC "
                        << packet_information.remote_ssrc;
      } else {
        LOG(LS_VERBOSE) << "Incoming FIR from SSRC "
                        << packet_information.remote_ssrc;
      }
      rtcp_intra_frame_observer_->OnReceivedIntraFrameRequest(local_ssrc);
    }
  }
  if (rtcp_bandwidth_observer_) {
    RTC_DCHECK(!receiver_only_);
    if (packet_information.packet_type_flags & kRtcpRemb) {
      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 == local_ssrc ||
        registered_ssrcs.find(media_source_ssrc) != registered_ssrcs.end()) {
      transport_feedback_observer_->OnTransportFeedback(
          *packet_information.transport_feedback);
    }
  }

  if (bitrate_allocation_observer_ &&
      packet_information.target_bitrate_allocation) {
    bitrate_allocation_observer_->OnBitrateAllocationUpdated(
        *packet_information.target_bitrate_allocation);
  }

  if (!receiver_only_) {
    rtc::CritScope cs(&feedbacks_lock_);
    if (stats_callback_) {
      for (const auto& report_block : packet_information.report_blocks) {
        RtcpStatistics stats;
        stats.cumulative_lost = report_block.cumulativeLost;
        stats.extended_max_sequence_number = report_block.extendedHighSeqNum;
        stats.fraction_lost = report_block.fractionLost;
        stats.jitter = report_block.jitter;

        stats_callback_->StatisticsUpdated(stats, report_block.sourceSSRC);
      }
    }
  }
}

int32_t RTCPReceiver::CNAME(uint32_t remoteSSRC,
                            char cName[RTCP_CNAME_SIZE]) const {
  RTC_DCHECK(cName);

  rtc::CritScope lock(&rtcp_receiver_lock_);
  auto received_cname_it = received_cnames_.find(remoteSSRC);
  if (received_cname_it == received_cnames_.end())
    return -1;

  size_t length = received_cname_it->second.copy(cName, RTCP_CNAME_SIZE - 1);
  cName[length] = 0;
  return 0;
}

std::vector<rtcp::TmmbItem> RTCPReceiver::TmmbrReceived() {
  rtc::CritScope lock(&rtcp_receiver_lock_);
  std::vector<rtcp::TmmbItem> candidates;

  int64_t now_ms = clock_->TimeInMilliseconds();
  // Use audio define since we don't know what interval the remote peer use.
  int64_t timeout_ms = now_ms - 5 * RTCP_INTERVAL_AUDIO_MS;

  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;
}

}  // namespace webrtc