/*
 *  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_sender.h"

#include <string.h>  // memcpy

#include <algorithm>  // std::min
#include <memory>
#include <utility>

#include "absl/types/optional.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/rtp_headers.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h"
#include "modules/rtp_rtcp/source/rtcp_packet/app.h"
#include "modules/rtp_rtcp/source/rtcp_packet/bye.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/receiver_report.h"
#include "modules/rtp_rtcp/source/rtcp_packet/remb.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_impl2.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_interface.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/numerics/safe_conversions.h"
#include "rtc_base/trace_event.h"

namespace webrtc {

namespace {
const uint32_t kRtcpAnyExtendedReports = kRtcpXrReceiverReferenceTime |
                                         kRtcpXrDlrrReportBlock |
                                         kRtcpXrTargetBitrate;
constexpr int32_t kDefaultVideoReportInterval = 1000;
constexpr int32_t kDefaultAudioReportInterval = 5000;
}  // namespace

// Helper to put several RTCP packets into lower layer datagram RTCP packet.
class RTCPSender::PacketSender {
 public:
  PacketSender(rtcp::RtcpPacket::PacketReadyCallback callback,
               size_t max_packet_size)
      : callback_(callback), max_packet_size_(max_packet_size) {
    RTC_CHECK_LE(max_packet_size, IP_PACKET_SIZE);
  }
  ~PacketSender() { RTC_DCHECK_EQ(index_, 0) << "Unsent rtcp packet."; }

  // Appends a packet to pending compound packet.
  // Sends rtcp packet if buffer is full and resets the buffer.
  void AppendPacket(const rtcp::RtcpPacket& packet) {
    packet.Create(buffer_, &index_, max_packet_size_, callback_);
  }

  // Sends pending rtcp packet.
  void Send() {
    if (index_ > 0) {
      callback_(rtc::ArrayView<const uint8_t>(buffer_, index_));
      index_ = 0;
    }
  }

 private:
  const rtcp::RtcpPacket::PacketReadyCallback callback_;
  const size_t max_packet_size_;
  size_t index_ = 0;
  uint8_t buffer_[IP_PACKET_SIZE];
};

RTCPSender::FeedbackState::FeedbackState()
    : packets_sent(0),
      media_bytes_sent(0),
      send_bitrate(0),
      last_rr_ntp_secs(0),
      last_rr_ntp_frac(0),
      remote_sr(0),
      receiver(nullptr) {}

RTCPSender::FeedbackState::FeedbackState(const FeedbackState&) = default;

RTCPSender::FeedbackState::FeedbackState(FeedbackState&&) = default;

RTCPSender::FeedbackState::~FeedbackState() = default;

class RTCPSender::RtcpContext {
 public:
  RtcpContext(const FeedbackState& feedback_state,
              int32_t nack_size,
              const uint16_t* nack_list,
              Timestamp now)
      : feedback_state_(feedback_state),
        nack_size_(nack_size),
        nack_list_(nack_list),
        now_(now) {}

  const FeedbackState& feedback_state_;
  const int32_t nack_size_;
  const uint16_t* nack_list_;
  const Timestamp now_;
};

RTCPSender::Configuration RTCPSender::Configuration::FromRtpRtcpConfiguration(
    const RtpRtcpInterface::Configuration& configuration) {
  RTCPSender::Configuration result;
  result.audio = configuration.audio;
  result.local_media_ssrc = configuration.local_media_ssrc;
  result.clock = configuration.clock;
  result.outgoing_transport = configuration.outgoing_transport;
  result.non_sender_rtt_measurement = configuration.non_sender_rtt_measurement;
  result.event_log = configuration.event_log;
  if (configuration.rtcp_report_interval_ms) {
    result.rtcp_report_interval =
        TimeDelta::Millis(configuration.rtcp_report_interval_ms);
  }
  result.receive_statistics = configuration.receive_statistics;
  result.rtcp_packet_type_counter_observer =
      configuration.rtcp_packet_type_counter_observer;
  return result;
}

RTCPSender::RTCPSender(Configuration config)
    : audio_(config.audio),
      ssrc_(config.local_media_ssrc),
      clock_(config.clock),
      random_(clock_->TimeInMicroseconds()),
      method_(RtcpMode::kOff),
      event_log_(config.event_log),
      transport_(config.outgoing_transport),
      report_interval_(config.rtcp_report_interval.value_or(
          TimeDelta::Millis(config.audio ? kDefaultAudioReportInterval
                                         : kDefaultVideoReportInterval))),
      schedule_next_rtcp_send_evaluation_function_(
          std::move(config.schedule_next_rtcp_send_evaluation_function)),
      sending_(false),
      timestamp_offset_(0),
      last_rtp_timestamp_(0),
      remote_ssrc_(0),
      receive_statistics_(config.receive_statistics),

      sequence_number_fir_(0),

      remb_bitrate_(0),

      tmmbr_send_bps_(0),
      packet_oh_send_(0),
      max_packet_size_(IP_PACKET_SIZE - 28),  // IPv4 + UDP by default.

      xr_send_receiver_reference_time_enabled_(
          config.non_sender_rtt_measurement),
      packet_type_counter_observer_(config.rtcp_packet_type_counter_observer),
      send_video_bitrate_allocation_(false),
      last_payload_type_(-1) {
  RTC_DCHECK(transport_ != nullptr);

  builders_[kRtcpSr] = &RTCPSender::BuildSR;
  builders_[kRtcpRr] = &RTCPSender::BuildRR;
  builders_[kRtcpSdes] = &RTCPSender::BuildSDES;
  builders_[kRtcpPli] = &RTCPSender::BuildPLI;
  builders_[kRtcpFir] = &RTCPSender::BuildFIR;
  builders_[kRtcpRemb] = &RTCPSender::BuildREMB;
  builders_[kRtcpBye] = &RTCPSender::BuildBYE;
  builders_[kRtcpLossNotification] = &RTCPSender::BuildLossNotification;
  builders_[kRtcpTmmbr] = &RTCPSender::BuildTMMBR;
  builders_[kRtcpTmmbn] = &RTCPSender::BuildTMMBN;
  builders_[kRtcpNack] = &RTCPSender::BuildNACK;
  builders_[kRtcpAnyExtendedReports] = &RTCPSender::BuildExtendedReports;
}

RTCPSender::~RTCPSender() {}

RtcpMode RTCPSender::Status() const {
  MutexLock lock(&mutex_rtcp_sender_);
  return method_;
}

void RTCPSender::SetRTCPStatus(RtcpMode new_method) {
  MutexLock lock(&mutex_rtcp_sender_);

  if (new_method == RtcpMode::kOff) {
    next_time_to_send_rtcp_ = absl::nullopt;
  } else if (method_ == RtcpMode::kOff) {
    // When switching on, reschedule the next packet
    SetNextRtcpSendEvaluationDuration(report_interval_ / 2);
  }
  method_ = new_method;
}

bool RTCPSender::Sending() const {
  MutexLock lock(&mutex_rtcp_sender_);
  return sending_;
}

void RTCPSender::SetSendingStatus(const FeedbackState& feedback_state,
                                  bool sending) {
  bool sendRTCPBye = false;
  {
    MutexLock lock(&mutex_rtcp_sender_);

    if (method_ != RtcpMode::kOff) {
      if (sending == false && sending_ == true) {
        // Trigger RTCP bye
        sendRTCPBye = true;
      }
    }
    sending_ = sending;
  }
  if (sendRTCPBye) {
    if (SendRTCP(feedback_state, kRtcpBye) != 0) {
      RTC_LOG(LS_WARNING) << "Failed to send RTCP BYE";
    }
  }
}

void RTCPSender::SetNonSenderRttMeasurement(bool enabled) {
  MutexLock lock(&mutex_rtcp_sender_);
  xr_send_receiver_reference_time_enabled_ = enabled;
}

int32_t RTCPSender::SendLossNotification(const FeedbackState& feedback_state,
                                         uint16_t last_decoded_seq_num,
                                         uint16_t last_received_seq_num,
                                         bool decodability_flag,
                                         bool buffering_allowed) {
  int32_t error_code = -1;
  auto callback = [&](rtc::ArrayView<const uint8_t> packet) {
    transport_->SendRtcp(packet.data(), packet.size());
    error_code = 0;
    if (event_log_) {
      event_log_->Log(std::make_unique<RtcEventRtcpPacketOutgoing>(packet));
    }
  };
  absl::optional<PacketSender> sender;
  {
    MutexLock lock(&mutex_rtcp_sender_);

    if (!loss_notification_.Set(last_decoded_seq_num, last_received_seq_num,
                                decodability_flag)) {
      return -1;
    }

    SetFlag(kRtcpLossNotification, /*is_volatile=*/true);

    if (buffering_allowed) {
      // The loss notification will be batched with additional feedback
      // messages.
      return 0;
    }

    sender.emplace(callback, max_packet_size_);
    auto result = ComputeCompoundRTCPPacket(
        feedback_state, RTCPPacketType::kRtcpLossNotification, 0, nullptr,
        *sender);
    if (result) {
      return *result;
    }
  }
  sender->Send();

  return error_code;
}

void RTCPSender::SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) {
  RTC_CHECK_GE(bitrate_bps, 0);
  MutexLock lock(&mutex_rtcp_sender_);
  if (method_ == RtcpMode::kOff) {
    RTC_LOG(LS_WARNING) << "Can't send rtcp if it is disabled.";
    return;
  }
  remb_bitrate_ = bitrate_bps;
  remb_ssrcs_ = std::move(ssrcs);

  SetFlag(kRtcpRemb, /*is_volatile=*/false);
  // Send a REMB immediately if we have a new REMB. The frequency of REMBs is
  // throttled by the caller.
  SetNextRtcpSendEvaluationDuration(TimeDelta::Zero());
}

void RTCPSender::UnsetRemb() {
  MutexLock lock(&mutex_rtcp_sender_);
  // Stop sending REMB each report until it is reenabled and REMB data set.
  ConsumeFlag(kRtcpRemb, /*forced=*/true);
}

bool RTCPSender::TMMBR() const {
  MutexLock lock(&mutex_rtcp_sender_);
  return IsFlagPresent(RTCPPacketType::kRtcpTmmbr);
}

void RTCPSender::SetMaxRtpPacketSize(size_t max_packet_size) {
  MutexLock lock(&mutex_rtcp_sender_);
  max_packet_size_ = max_packet_size;
}

void RTCPSender::SetTimestampOffset(uint32_t timestamp_offset) {
  MutexLock lock(&mutex_rtcp_sender_);
  timestamp_offset_ = timestamp_offset;
}

void RTCPSender::SetLastRtpTime(uint32_t rtp_timestamp,
                                absl::optional<Timestamp> capture_time,
                                absl::optional<int8_t> payload_type) {
  MutexLock lock(&mutex_rtcp_sender_);
  // For compatibility with clients who don't set payload type correctly on all
  // calls.
  if (payload_type.has_value()) {
    last_payload_type_ = *payload_type;
  }
  last_rtp_timestamp_ = rtp_timestamp;
  if (!capture_time.has_value()) {
    // We don't currently get a capture time from VoiceEngine.
    last_frame_capture_time_ = clock_->CurrentTime();
  } else {
    last_frame_capture_time_ = *capture_time;
  }
}

void RTCPSender::SetRtpClockRate(int8_t payload_type, int rtp_clock_rate_hz) {
  MutexLock lock(&mutex_rtcp_sender_);
  rtp_clock_rates_khz_[payload_type] = rtp_clock_rate_hz / 1000;
}

uint32_t RTCPSender::SSRC() const {
  MutexLock lock(&mutex_rtcp_sender_);
  return ssrc_;
}

void RTCPSender::SetSsrc(uint32_t ssrc) {
  MutexLock lock(&mutex_rtcp_sender_);
  ssrc_ = ssrc;
}

void RTCPSender::SetRemoteSSRC(uint32_t ssrc) {
  MutexLock lock(&mutex_rtcp_sender_);
  remote_ssrc_ = ssrc;
}

int32_t RTCPSender::SetCNAME(const char* c_name) {
  if (!c_name)
    return -1;

  RTC_DCHECK_LT(strlen(c_name), RTCP_CNAME_SIZE);
  MutexLock lock(&mutex_rtcp_sender_);
  cname_ = c_name;
  return 0;
}

bool RTCPSender::TimeToSendRTCPReport(bool sendKeyframeBeforeRTP) const {
  /*
      For audio we use a configurable interval (default: 5 seconds)

      For video we use a configurable interval (default: 1 second) for a BW
          smaller than 360 kbit/s, technicaly we break the max 5% RTCP BW for
          video below 10 kbit/s but that should be extremely rare


  From RFC 3550

      MAX RTCP BW is 5% if the session BW
          A send report is approximately 65 bytes inc CNAME
          A receiver report is approximately 28 bytes

      The RECOMMENDED value for the reduced minimum in seconds is 360
        divided by the session bandwidth in kilobits/second.  This minimum
        is smaller than 5 seconds for bandwidths greater than 72 kb/s.

      If the participant has not yet sent an RTCP packet (the variable
        initial is true), the constant Tmin is set to half of the configured
        interval.

      The interval between RTCP packets is varied randomly over the
        range [0.5,1.5] times the calculated interval to avoid unintended
        synchronization of all participants

      if we send
      If the participant is a sender (we_sent true), the constant C is
        set to the average RTCP packet size (avg_rtcp_size) divided by 25%
        of the RTCP bandwidth (rtcp_bw), and the constant n is set to the
        number of senders.

      if we receive only
        If we_sent is not true, the constant C is set
        to the average RTCP packet size divided by 75% of the RTCP
        bandwidth.  The constant n is set to the number of receivers
        (members - senders).  If the number of senders is greater than
        25%, senders and receivers are treated together.

      reconsideration NOT required for peer-to-peer
        "timer reconsideration" is
        employed.  This algorithm implements a simple back-off mechanism
        which causes users to hold back RTCP packet transmission if the
        group sizes are increasing.

        n = number of members
        C = avg_size/(rtcpBW/4)

     3. The deterministic calculated interval Td is set to max(Tmin, n*C).

     4. The calculated interval T is set to a number uniformly distributed
        between 0.5 and 1.5 times the deterministic calculated interval.

     5. The resulting value of T is divided by e-3/2=1.21828 to compensate
        for the fact that the timer reconsideration algorithm converges to
        a value of the RTCP bandwidth below the intended average
  */

  Timestamp now = clock_->CurrentTime();

  MutexLock lock(&mutex_rtcp_sender_);
  RTC_DCHECK(
      (method_ == RtcpMode::kOff && !next_time_to_send_rtcp_.has_value()) ||
      (method_ != RtcpMode::kOff && next_time_to_send_rtcp_.has_value()));
  if (method_ == RtcpMode::kOff)
    return false;

  if (!audio_ && sendKeyframeBeforeRTP) {
    // for video key-frames we want to send the RTCP before the large key-frame
    // if we have a 100 ms margin
    now += RTCP_SEND_BEFORE_KEY_FRAME;
  }

  return now >= *next_time_to_send_rtcp_;
}

void RTCPSender::BuildSR(const RtcpContext& ctx, PacketSender& sender) {
  // Timestamp shouldn't be estimated before first media frame.
  RTC_DCHECK(last_frame_capture_time_.has_value());
  // The timestamp of this RTCP packet should be estimated as the timestamp of
  // the frame being captured at this moment. We are calculating that
  // timestamp as the last frame's timestamp + the time since the last frame
  // was captured.
  int rtp_rate = rtp_clock_rates_khz_[last_payload_type_];
  if (rtp_rate <= 0) {
    rtp_rate =
        (audio_ ? kBogusRtpRateForAudioRtcp : kVideoPayloadTypeFrequency) /
        1000;
  }
  // Round now_us_ to the closest millisecond, because Ntp time is rounded
  // when converted to milliseconds,
  uint32_t rtp_timestamp =
      timestamp_offset_ + last_rtp_timestamp_ +
      ((ctx.now_.us() + 500) / 1000 - last_frame_capture_time_->ms()) *
          rtp_rate;

  rtcp::SenderReport report;
  report.SetSenderSsrc(ssrc_);
  report.SetNtp(clock_->ConvertTimestampToNtpTime(ctx.now_));
  report.SetRtpTimestamp(rtp_timestamp);
  report.SetPacketCount(ctx.feedback_state_.packets_sent);
  report.SetOctetCount(ctx.feedback_state_.media_bytes_sent);
  report.SetReportBlocks(CreateReportBlocks(ctx.feedback_state_));
  sender.AppendPacket(report);
}

void RTCPSender::BuildSDES(const RtcpContext& ctx, PacketSender& sender) {
  size_t length_cname = cname_.length();
  RTC_CHECK_LT(length_cname, RTCP_CNAME_SIZE);

  rtcp::Sdes sdes;
  sdes.AddCName(ssrc_, cname_);
  sender.AppendPacket(sdes);
}

void RTCPSender::BuildRR(const RtcpContext& ctx, PacketSender& sender) {
  rtcp::ReceiverReport report;
  report.SetSenderSsrc(ssrc_);
  report.SetReportBlocks(CreateReportBlocks(ctx.feedback_state_));
  sender.AppendPacket(report);
}

void RTCPSender::BuildPLI(const RtcpContext& ctx, PacketSender& sender) {
  rtcp::Pli pli;
  pli.SetSenderSsrc(ssrc_);
  pli.SetMediaSsrc(remote_ssrc_);

  ++packet_type_counter_.pli_packets;
  sender.AppendPacket(pli);
}

void RTCPSender::BuildFIR(const RtcpContext& ctx, PacketSender& sender) {
  ++sequence_number_fir_;

  rtcp::Fir fir;
  fir.SetSenderSsrc(ssrc_);
  fir.AddRequestTo(remote_ssrc_, sequence_number_fir_);

  ++packet_type_counter_.fir_packets;
  sender.AppendPacket(fir);
}

void RTCPSender::BuildREMB(const RtcpContext& ctx, PacketSender& sender) {
  rtcp::Remb remb;
  remb.SetSenderSsrc(ssrc_);
  remb.SetBitrateBps(remb_bitrate_);
  remb.SetSsrcs(remb_ssrcs_);
  sender.AppendPacket(remb);
}

void RTCPSender::SetTargetBitrate(unsigned int target_bitrate) {
  MutexLock lock(&mutex_rtcp_sender_);
  tmmbr_send_bps_ = target_bitrate;
}

void RTCPSender::BuildTMMBR(const RtcpContext& ctx, PacketSender& sender) {
  if (ctx.feedback_state_.receiver == nullptr)
    return;
  // Before sending the TMMBR check the received TMMBN, only an owner is
  // allowed to raise the bitrate:
  // * If the sender is an owner of the TMMBN -> send TMMBR
  // * If not an owner but the TMMBR would enter the TMMBN -> send TMMBR

  // get current bounding set from RTCP receiver
  bool tmmbr_owner = false;

  // holding mutex_rtcp_sender_ while calling RTCPreceiver which
  // will accuire criticalSectionRTCPReceiver_ is a potental deadlock but
  // since RTCPreceiver is not doing the reverse we should be fine
  std::vector<rtcp::TmmbItem> candidates =
      ctx.feedback_state_.receiver->BoundingSet(&tmmbr_owner);

  if (!candidates.empty()) {
    for (const auto& candidate : candidates) {
      if (candidate.bitrate_bps() == tmmbr_send_bps_ &&
          candidate.packet_overhead() == packet_oh_send_) {
        // Do not send the same tuple.
        return;
      }
    }
    if (!tmmbr_owner) {
      // Use received bounding set as candidate set.
      // Add current tuple.
      candidates.emplace_back(ssrc_, tmmbr_send_bps_, packet_oh_send_);

      // Find bounding set.
      std::vector<rtcp::TmmbItem> bounding =
          TMMBRHelp::FindBoundingSet(std::move(candidates));
      tmmbr_owner = TMMBRHelp::IsOwner(bounding, ssrc_);
      if (!tmmbr_owner) {
        // Did not enter bounding set, no meaning to send this request.
        return;
      }
    }
  }

  if (!tmmbr_send_bps_)
    return;

  rtcp::Tmmbr tmmbr;
  tmmbr.SetSenderSsrc(ssrc_);
  rtcp::TmmbItem request;
  request.set_ssrc(remote_ssrc_);
  request.set_bitrate_bps(tmmbr_send_bps_);
  request.set_packet_overhead(packet_oh_send_);
  tmmbr.AddTmmbr(request);
  sender.AppendPacket(tmmbr);
}

void RTCPSender::BuildTMMBN(const RtcpContext& ctx, PacketSender& sender) {
  rtcp::Tmmbn tmmbn;
  tmmbn.SetSenderSsrc(ssrc_);
  for (const rtcp::TmmbItem& tmmbr : tmmbn_to_send_) {
    if (tmmbr.bitrate_bps() > 0) {
      tmmbn.AddTmmbr(tmmbr);
    }
  }
  sender.AppendPacket(tmmbn);
}

void RTCPSender::BuildAPP(const RtcpContext& ctx, PacketSender& sender) {
  rtcp::App app;
  app.SetSenderSsrc(ssrc_);
  sender.AppendPacket(app);
}

void RTCPSender::BuildLossNotification(const RtcpContext& ctx,
                                       PacketSender& sender) {
  loss_notification_.SetSenderSsrc(ssrc_);
  loss_notification_.SetMediaSsrc(remote_ssrc_);
  sender.AppendPacket(loss_notification_);
}

void RTCPSender::BuildNACK(const RtcpContext& ctx, PacketSender& sender) {
  rtcp::Nack nack;
  nack.SetSenderSsrc(ssrc_);
  nack.SetMediaSsrc(remote_ssrc_);
  nack.SetPacketIds(ctx.nack_list_, ctx.nack_size_);

  // Report stats.
  for (int idx = 0; idx < ctx.nack_size_; ++idx) {
    nack_stats_.ReportRequest(ctx.nack_list_[idx]);
  }
  packet_type_counter_.nack_requests = nack_stats_.requests();
  packet_type_counter_.unique_nack_requests = nack_stats_.unique_requests();

  ++packet_type_counter_.nack_packets;
  sender.AppendPacket(nack);
}

void RTCPSender::BuildBYE(const RtcpContext& ctx, PacketSender& sender) {
  rtcp::Bye bye;
  bye.SetSenderSsrc(ssrc_);
  bye.SetCsrcs(csrcs_);
  sender.AppendPacket(bye);
}

void RTCPSender::BuildExtendedReports(const RtcpContext& ctx,
                                      PacketSender& sender) {
  rtcp::ExtendedReports xr;
  xr.SetSenderSsrc(ssrc_);

  if (!sending_ && xr_send_receiver_reference_time_enabled_) {
    rtcp::Rrtr rrtr;
    rrtr.SetNtp(clock_->ConvertTimestampToNtpTime(ctx.now_));
    xr.SetRrtr(rrtr);
  }

  for (const rtcp::ReceiveTimeInfo& rti : ctx.feedback_state_.last_xr_rtis) {
    xr.AddDlrrItem(rti);
  }

  if (send_video_bitrate_allocation_) {
    rtcp::TargetBitrate target_bitrate;

    for (int sl = 0; sl < kMaxSpatialLayers; ++sl) {
      for (int tl = 0; tl < kMaxTemporalStreams; ++tl) {
        if (video_bitrate_allocation_.HasBitrate(sl, tl)) {
          target_bitrate.AddTargetBitrate(
              sl, tl, video_bitrate_allocation_.GetBitrate(sl, tl) / 1000);
        }
      }
    }

    xr.SetTargetBitrate(target_bitrate);
    send_video_bitrate_allocation_ = false;
  }
  sender.AppendPacket(xr);
}

int32_t RTCPSender::SendRTCP(const FeedbackState& feedback_state,
                             RTCPPacketType packet_type,
                             int32_t nack_size,
                             const uint16_t* nack_list) {
  int32_t error_code = -1;
  auto callback = [&](rtc::ArrayView<const uint8_t> packet) {
    if (transport_->SendRtcp(packet.data(), packet.size())) {
      error_code = 0;
      if (event_log_) {
        event_log_->Log(std::make_unique<RtcEventRtcpPacketOutgoing>(packet));
      }
    }
  };
  absl::optional<PacketSender> sender;
  {
    MutexLock lock(&mutex_rtcp_sender_);
    sender.emplace(callback, max_packet_size_);
    auto result = ComputeCompoundRTCPPacket(feedback_state, packet_type,
                                            nack_size, nack_list, *sender);
    if (result) {
      return *result;
    }
  }
  sender->Send();

  return error_code;
}

absl::optional<int32_t> RTCPSender::ComputeCompoundRTCPPacket(
    const FeedbackState& feedback_state,
    RTCPPacketType packet_type,
    int32_t nack_size,
    const uint16_t* nack_list,
    PacketSender& sender) {
  if (method_ == RtcpMode::kOff) {
    RTC_LOG(LS_WARNING) << "Can't send rtcp if it is disabled.";
    return -1;
  }
  // Add the flag as volatile. Non volatile entries will not be overwritten.
  // The new volatile flag will be consumed by the end of this call.
  SetFlag(packet_type, true);

  // Prevent sending streams to send SR before any media has been sent.
  const bool can_calculate_rtp_timestamp = last_frame_capture_time_.has_value();
  if (!can_calculate_rtp_timestamp) {
    bool consumed_sr_flag = ConsumeFlag(kRtcpSr);
    bool consumed_report_flag = sending_ && ConsumeFlag(kRtcpReport);
    bool sender_report = consumed_report_flag || consumed_sr_flag;
    if (sender_report && AllVolatileFlagsConsumed()) {
      // This call was for Sender Report and nothing else.
      return 0;
    }
    if (sending_ && method_ == RtcpMode::kCompound) {
      // Not allowed to send any RTCP packet without sender report.
      return -1;
    }
  }

  if (packet_type_counter_.first_packet_time_ms == -1)
    packet_type_counter_.first_packet_time_ms = clock_->TimeInMilliseconds();

  // We need to send our NTP even if we haven't received any reports.
  RtcpContext context(feedback_state, nack_size, nack_list,
                      clock_->CurrentTime());

  PrepareReport(feedback_state);

  bool create_bye = false;

  auto it = report_flags_.begin();
  while (it != report_flags_.end()) {
    uint32_t rtcp_packet_type = it->type;

    if (it->is_volatile) {
      report_flags_.erase(it++);
    } else {
      ++it;
    }

    // If there is a BYE, don't append now - save it and append it
    // at the end later.
    if (rtcp_packet_type == kRtcpBye) {
      create_bye = true;
      continue;
    }
    auto builder_it = builders_.find(rtcp_packet_type);
    if (builder_it == builders_.end()) {
      RTC_NOTREACHED() << "Could not find builder for packet type "
                       << rtcp_packet_type;
    } else {
      BuilderFunc func = builder_it->second;
      (this->*func)(context, sender);
    }
  }

  // Append the BYE now at the end
  if (create_bye) {
    BuildBYE(context, sender);
  }

  if (packet_type_counter_observer_ != nullptr) {
    packet_type_counter_observer_->RtcpPacketTypesCounterUpdated(
        remote_ssrc_, packet_type_counter_);
  }

  RTC_DCHECK(AllVolatileFlagsConsumed());
  return absl::nullopt;
}

void RTCPSender::PrepareReport(const FeedbackState& feedback_state) {
  bool generate_report;
  if (IsFlagPresent(kRtcpSr) || IsFlagPresent(kRtcpRr)) {
    // Report type already explicitly set, don't automatically populate.
    generate_report = true;
    RTC_DCHECK(ConsumeFlag(kRtcpReport) == false);
  } else {
    generate_report =
        (ConsumeFlag(kRtcpReport) && method_ == RtcpMode::kReducedSize) ||
        method_ == RtcpMode::kCompound;
    if (generate_report)
      SetFlag(sending_ ? kRtcpSr : kRtcpRr, true);
  }

  if (IsFlagPresent(kRtcpSr) || (IsFlagPresent(kRtcpRr) && !cname_.empty()))
    SetFlag(kRtcpSdes, true);

  if (generate_report) {
    if ((!sending_ && xr_send_receiver_reference_time_enabled_) ||
        !feedback_state.last_xr_rtis.empty() ||
        send_video_bitrate_allocation_) {
      SetFlag(kRtcpAnyExtendedReports, true);
    }

    // generate next time to send an RTCP report
    TimeDelta min_interval = report_interval_;

    if (!audio_ && sending_) {
      // Calculate bandwidth for video; 360 / send bandwidth in kbit/s.
      int send_bitrate_kbit = feedback_state.send_bitrate / 1000;
      if (send_bitrate_kbit != 0) {
        min_interval = std::min(TimeDelta::Millis(360000 / send_bitrate_kbit),
                                report_interval_);
      }
    }

    // The interval between RTCP packets is varied randomly over the
    // range [1/2,3/2] times the calculated interval.
    int min_interval_int = rtc::dchecked_cast<int>(min_interval.ms());
    TimeDelta time_to_next = TimeDelta::Millis(
        random_.Rand(min_interval_int * 1 / 2, min_interval_int * 3 / 2));

    RTC_DCHECK(!time_to_next.IsZero());
    SetNextRtcpSendEvaluationDuration(time_to_next);

    // RtcpSender expected to be used for sending either just sender reports
    // or just receiver reports.
    RTC_DCHECK(!(IsFlagPresent(kRtcpSr) && IsFlagPresent(kRtcpRr)));
  }
}

std::vector<rtcp::ReportBlock> RTCPSender::CreateReportBlocks(
    const FeedbackState& feedback_state) {
  std::vector<rtcp::ReportBlock> result;
  if (!receive_statistics_)
    return result;

  // TODO(danilchap): Support sending more than `RTCP_MAX_REPORT_BLOCKS` per
  // compound rtcp packet when single rtcp module is used for multiple media
  // streams.
  result = receive_statistics_->RtcpReportBlocks(RTCP_MAX_REPORT_BLOCKS);

  if (!result.empty() && ((feedback_state.last_rr_ntp_secs != 0) ||
                          (feedback_state.last_rr_ntp_frac != 0))) {
    // Get our NTP as late as possible to avoid a race.
    uint32_t now = CompactNtp(clock_->CurrentNtpTime());

    uint32_t receive_time = feedback_state.last_rr_ntp_secs & 0x0000FFFF;
    receive_time <<= 16;
    receive_time += (feedback_state.last_rr_ntp_frac & 0xffff0000) >> 16;

    uint32_t delay_since_last_sr = now - receive_time;
    // TODO(danilchap): Instead of setting same value on all report blocks,
    // set only when media_ssrc match sender ssrc of the sender report
    // remote times were taken from.
    for (auto& report_block : result) {
      report_block.SetLastSr(feedback_state.remote_sr);
      report_block.SetDelayLastSr(delay_since_last_sr);
    }
  }
  return result;
}

void RTCPSender::SetCsrcs(const std::vector<uint32_t>& csrcs) {
  RTC_DCHECK_LE(csrcs.size(), kRtpCsrcSize);
  MutexLock lock(&mutex_rtcp_sender_);
  csrcs_ = csrcs;
}

void RTCPSender::SetTmmbn(std::vector<rtcp::TmmbItem> bounding_set) {
  MutexLock lock(&mutex_rtcp_sender_);
  tmmbn_to_send_ = std::move(bounding_set);
  SetFlag(kRtcpTmmbn, true);
}

void RTCPSender::SetFlag(uint32_t type, bool is_volatile) {
  if (type & kRtcpAnyExtendedReports) {
    report_flags_.insert(ReportFlag(kRtcpAnyExtendedReports, is_volatile));
  } else {
    report_flags_.insert(ReportFlag(type, is_volatile));
  }
}

bool RTCPSender::IsFlagPresent(uint32_t type) const {
  return report_flags_.find(ReportFlag(type, false)) != report_flags_.end();
}

bool RTCPSender::ConsumeFlag(uint32_t type, bool forced) {
  auto it = report_flags_.find(ReportFlag(type, false));
  if (it == report_flags_.end())
    return false;
  if (it->is_volatile || forced)
    report_flags_.erase((it));
  return true;
}

bool RTCPSender::AllVolatileFlagsConsumed() const {
  for (const ReportFlag& flag : report_flags_) {
    if (flag.is_volatile)
      return false;
  }
  return true;
}

void RTCPSender::SetVideoBitrateAllocation(
    const VideoBitrateAllocation& bitrate) {
  MutexLock lock(&mutex_rtcp_sender_);
  if (method_ == RtcpMode::kOff) {
    RTC_LOG(LS_WARNING) << "Can't send rtcp if it is disabled.";
    return;
  }
  // Check if this allocation is first ever, or has a different set of
  // spatial/temporal layers signaled and enabled, if so trigger an rtcp report
  // as soon as possible.
  absl::optional<VideoBitrateAllocation> new_bitrate =
      CheckAndUpdateLayerStructure(bitrate);
  if (new_bitrate) {
    video_bitrate_allocation_ = *new_bitrate;
    RTC_LOG(LS_INFO) << "Emitting TargetBitrate XR for SSRC " << ssrc_
                     << " with new layers enabled/disabled: "
                     << video_bitrate_allocation_.ToString();
    SetNextRtcpSendEvaluationDuration(TimeDelta::Zero());
  } else {
    video_bitrate_allocation_ = bitrate;
  }

  send_video_bitrate_allocation_ = true;
  SetFlag(kRtcpAnyExtendedReports, true);
}

absl::optional<VideoBitrateAllocation> RTCPSender::CheckAndUpdateLayerStructure(
    const VideoBitrateAllocation& bitrate) const {
  absl::optional<VideoBitrateAllocation> updated_bitrate;
  for (size_t si = 0; si < kMaxSpatialLayers; ++si) {
    for (size_t ti = 0; ti < kMaxTemporalStreams; ++ti) {
      if (!updated_bitrate &&
          (bitrate.HasBitrate(si, ti) !=
               video_bitrate_allocation_.HasBitrate(si, ti) ||
           (bitrate.GetBitrate(si, ti) == 0) !=
               (video_bitrate_allocation_.GetBitrate(si, ti) == 0))) {
        updated_bitrate = bitrate;
      }
      if (video_bitrate_allocation_.GetBitrate(si, ti) > 0 &&
          bitrate.GetBitrate(si, ti) == 0) {
        // Make sure this stream disabling is explicitly signaled.
        updated_bitrate->SetBitrate(si, ti, 0);
      }
    }
  }

  return updated_bitrate;
}

void RTCPSender::SendCombinedRtcpPacket(
    std::vector<std::unique_ptr<rtcp::RtcpPacket>> rtcp_packets) {
  size_t max_packet_size;
  uint32_t ssrc;
  {
    MutexLock lock(&mutex_rtcp_sender_);
    if (method_ == RtcpMode::kOff) {
      RTC_LOG(LS_WARNING) << "Can't send rtcp if it is disabled.";
      return;
    }

    max_packet_size = max_packet_size_;
    ssrc = ssrc_;
  }
  RTC_DCHECK_LE(max_packet_size, IP_PACKET_SIZE);
  auto callback = [&](rtc::ArrayView<const uint8_t> packet) {
    if (transport_->SendRtcp(packet.data(), packet.size())) {
      if (event_log_)
        event_log_->Log(std::make_unique<RtcEventRtcpPacketOutgoing>(packet));
    }
  };
  PacketSender sender(callback, max_packet_size);
  for (auto& rtcp_packet : rtcp_packets) {
    rtcp_packet->SetSenderSsrc(ssrc);
    sender.AppendPacket(*rtcp_packet);
  }
  sender.Send();
}

void RTCPSender::SetNextRtcpSendEvaluationDuration(TimeDelta duration) {
  next_time_to_send_rtcp_ = clock_->CurrentTime() + duration;
  // TODO(bugs.webrtc.org/11581): make unconditional once downstream consumers
  // are using the callback method.
  if (schedule_next_rtcp_send_evaluation_function_)
    schedule_next_rtcp_send_evaluation_function_(duration);
}

}  // namespace webrtc