modules/rtp_rtcp/source/rtcp_sender.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 "modules/rtp_rtcp/source/rtcp_sender.h"

#include <string.h>  // memcpy

#include <utility>

#include "common_types.h"  // NOLINT(build/include)
#include "logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h"
#include "logging/rtc_event_log/rtc_event_log.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/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_impl.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/constructormagic.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/trace_event.h"

namespace webrtc {

namespace {
const uint32_t kRtcpAnyExtendedReports =
    kRtcpXrVoipMetric | kRtcpXrReceiverReferenceTime | kRtcpXrDlrrReportBlock |
    kRtcpXrTargetBitrate;
}  // namespace

NACKStringBuilder::NACKStringBuilder()
    : stream_(""), count_(0), prevNack_(0), consecutive_(false) {}

NACKStringBuilder::~NACKStringBuilder() {}

void NACKStringBuilder::PushNACK(uint16_t nack) {
  if (count_ == 0) {
    stream_ << nack;
  } else if (nack == prevNack_ + 1) {
    consecutive_ = true;
  } else {
    if (consecutive_) {
      stream_ << "-" << prevNack_;
      consecutive_ = false;
    }
    stream_ << "," << nack;
  }
  count_++;
  prevNack_ = nack;
}

std::string NACKStringBuilder::GetResult() {
  if (consecutive_) {
    stream_ << "-" << prevNack_;
    consecutive_ = false;
  }
  return stream_.str();
}

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),
      module(nullptr) {}

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

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

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

class PacketContainer : public rtcp::CompoundPacket {
 public:
  PacketContainer(Transport* transport, RtcEventLog* event_log)
      : transport_(transport), event_log_(event_log) {}
  ~PacketContainer() override {
    for (RtcpPacket* packet : appended_packets_)
      delete packet;
  }

  size_t SendPackets(size_t max_payload_length) {
    size_t bytes_sent = 0;
    Build(max_payload_length, [&](rtc::ArrayView<const uint8_t> packet) {
      if (transport_->SendRtcp(packet.data(), packet.size())) {
        bytes_sent += packet.size();
        if (event_log_) {
          event_log_->Log(rtc::MakeUnique<RtcEventRtcpPacketOutgoing>(packet));
        }
      }
    });
    return bytes_sent;
  }

 private:
  Transport* transport_;
  RtcEventLog* const event_log_;

  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(PacketContainer);
};

class RTCPSender::RtcpContext {
 public:
  RtcpContext(const FeedbackState& feedback_state,
              int32_t nack_size,
              const uint16_t* nack_list,
              NtpTime 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 NtpTime now_;
};

RTCPSender::RTCPSender(
    bool audio,
    Clock* clock,
    ReceiveStatisticsProvider* receive_statistics,
    RtcpPacketTypeCounterObserver* packet_type_counter_observer,
    RtcEventLog* event_log,
    Transport* outgoing_transport,
    RtcpIntervalConfig interval_config)
    : audio_(audio),
      clock_(clock),
      random_(clock_->TimeInMicroseconds()),
      method_(RtcpMode::kOff),
      event_log_(event_log),
      transport_(outgoing_transport),
      interval_config_(interval_config),
      using_nack_(false),
      sending_(false),
      next_time_to_send_rtcp_(0),
      timestamp_offset_(0),
      last_rtp_timestamp_(0),
      last_frame_capture_time_ms_(-1),
      ssrc_(0),
      remote_ssrc_(0),
      receive_statistics_(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.

      app_sub_type_(0),
      app_name_(0),
      app_data_(nullptr),
      app_length_(0),

      xr_send_receiver_reference_time_enabled_(false),
      packet_type_counter_observer_(packet_type_counter_observer) {
  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_[kRtcpApp] = &RTCPSender::BuildAPP;
  builders_[kRtcpTmmbr] = &RTCPSender::BuildTMMBR;
  builders_[kRtcpTmmbn] = &RTCPSender::BuildTMMBN;
  builders_[kRtcpNack] = &RTCPSender::BuildNACK;
  builders_[kRtcpAnyExtendedReports] = &RTCPSender::BuildExtendedReports;
}

RTCPSender::~RTCPSender() {}

RtcpMode RTCPSender::Status() const {
  rtc::CritScope lock(&critical_section_rtcp_sender_);
  return method_;
}

void RTCPSender::SetRTCPStatus(RtcpMode new_method) {
  rtc::CritScope lock(&critical_section_rtcp_sender_);

  if (method_ == RtcpMode::kOff && new_method != RtcpMode::kOff) {
    // When switching on, reschedule the next packet
    int64_t interval_ms = audio_ ? interval_config_.audio_interval_ms
                                 : interval_config_.video_interval_ms;
    next_time_to_send_rtcp_ = clock_->TimeInMilliseconds() + (interval_ms / 2);
  }
  method_ = new_method;
}

bool RTCPSender::Sending() const {
  rtc::CritScope lock(&critical_section_rtcp_sender_);
  return sending_;
}

int32_t RTCPSender::SetSendingStatus(const FeedbackState& feedback_state,
                                     bool sending) {
  bool sendRTCPBye = false;
  {
    rtc::CritScope lock(&critical_section_rtcp_sender_);

    if (method_ != RtcpMode::kOff) {
      if (sending == false && sending_ == true) {
        // Trigger RTCP bye
        sendRTCPBye = true;
      }
    }
    sending_ = sending;
  }
  if (sendRTCPBye)
    return SendRTCP(feedback_state, kRtcpBye);
  return 0;
}

void RTCPSender::SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) {
  RTC_CHECK_GE(bitrate_bps, 0);
  rtc::CritScope lock(&critical_section_rtcp_sender_);
  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.
  next_time_to_send_rtcp_ = clock_->TimeInMilliseconds();
}

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

bool RTCPSender::TMMBR() const {
  rtc::CritScope lock(&critical_section_rtcp_sender_);
  return IsFlagPresent(RTCPPacketType::kRtcpTmmbr);
}

void RTCPSender::SetTMMBRStatus(bool enable) {
  rtc::CritScope lock(&critical_section_rtcp_sender_);
  if (enable) {
    SetFlag(RTCPPacketType::kRtcpTmmbr, false);
  } else {
    ConsumeFlag(RTCPPacketType::kRtcpTmmbr, true);
  }
}

void RTCPSender::SetMaxRtpPacketSize(size_t max_packet_size) {
  rtc::CritScope lock(&critical_section_rtcp_sender_);
  max_packet_size_ = max_packet_size;
}

void RTCPSender::SetTimestampOffset(uint32_t timestamp_offset) {
  rtc::CritScope lock(&critical_section_rtcp_sender_);
  timestamp_offset_ = timestamp_offset;
}

void RTCPSender::SetLastRtpTime(uint32_t rtp_timestamp,
                                int64_t capture_time_ms) {
  rtc::CritScope lock(&critical_section_rtcp_sender_);
  last_rtp_timestamp_ = rtp_timestamp;
  if (capture_time_ms < 0) {
    // We don't currently get a capture time from VoiceEngine.
    last_frame_capture_time_ms_ = clock_->TimeInMilliseconds();
  } else {
    last_frame_capture_time_ms_ = capture_time_ms;
  }
}

uint32_t RTCPSender::SSRC() const {
  rtc::CritScope lock(&critical_section_rtcp_sender_);
  return ssrc_;
}

void RTCPSender::SetSSRC(uint32_t ssrc) {
  rtc::CritScope lock(&critical_section_rtcp_sender_);

  if (ssrc_ != 0) {
    // not first SetSSRC, probably due to a collision
    // schedule a new RTCP report
    // make sure that we send a RTP packet
    next_time_to_send_rtcp_ = clock_->TimeInMilliseconds() + 100;
  }
  ssrc_ = ssrc;
}

void RTCPSender::SetRemoteSSRC(uint32_t ssrc) {
  rtc::CritScope lock(&critical_section_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);
  rtc::CritScope lock(&critical_section_rtcp_sender_);
  cname_ = c_name;
  return 0;
}

int32_t RTCPSender::AddMixedCNAME(uint32_t SSRC, const char* c_name) {
  RTC_DCHECK(c_name);
  RTC_DCHECK_LT(strlen(c_name), RTCP_CNAME_SIZE);
  rtc::CritScope lock(&critical_section_rtcp_sender_);
  // One spot is reserved for ssrc_/cname_.
  // TODO(danilchap): Add support for more than 30 contributes by sending
  // several sdes packets.
  if (csrc_cnames_.size() >= rtcp::Sdes::kMaxNumberOfChunks - 1)
    return -1;

  csrc_cnames_[SSRC] = c_name;
  return 0;
}

int32_t RTCPSender::RemoveMixedCNAME(uint32_t SSRC) {
  rtc::CritScope lock(&critical_section_rtcp_sender_);
  auto it = csrc_cnames_.find(SSRC);

  if (it == csrc_cnames_.end())
    return -1;

  csrc_cnames_.erase(it);
  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
  */

  int64_t now = clock_->TimeInMilliseconds();

  rtc::CritScope lock(&critical_section_rtcp_sender_);

  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_MS;
  }

  if (now >= next_time_to_send_rtcp_) {
    return true;
  } else if (now < 0x0000ffff &&
             next_time_to_send_rtcp_ > 0xffff0000) {  // 65 sec margin
    // wrap
    return true;
  }
  return false;
}

std::unique_ptr<rtcp::RtcpPacket> RTCPSender::BuildSR(const RtcpContext& ctx) {
  // Timestamp shouldn't be estimated before first media frame.
  RTC_DCHECK_GE(last_frame_capture_time_ms_, 0);
  // 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.
  uint32_t rtp_rate =
      (audio_ ? kBogusRtpRateForAudioRtcp : kVideoPayloadTypeFrequency) / 1000;
  uint32_t rtp_timestamp =
      timestamp_offset_ + last_rtp_timestamp_ +
      (clock_->TimeInMilliseconds() - last_frame_capture_time_ms_) * rtp_rate;

  rtcp::SenderReport* report = new rtcp::SenderReport();
  report->SetSenderSsrc(ssrc_);
  report->SetNtp(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_));

  return std::unique_ptr<rtcp::RtcpPacket>(report);
}

std::unique_ptr<rtcp::RtcpPacket> RTCPSender::BuildSDES(
    const RtcpContext& ctx) {
  size_t length_cname = cname_.length();
  RTC_CHECK_LT(length_cname, RTCP_CNAME_SIZE);

  rtcp::Sdes* sdes = new rtcp::Sdes();
  sdes->AddCName(ssrc_, cname_);

  for (const auto& it : csrc_cnames_)
    RTC_CHECK(sdes->AddCName(it.first, it.second));

  return std::unique_ptr<rtcp::RtcpPacket>(sdes);
}

std::unique_ptr<rtcp::RtcpPacket> RTCPSender::BuildRR(const RtcpContext& ctx) {
  rtcp::ReceiverReport* report = new rtcp::ReceiverReport();
  report->SetSenderSsrc(ssrc_);
  report->SetReportBlocks(CreateReportBlocks(ctx.feedback_state_));

  return std::unique_ptr<rtcp::RtcpPacket>(report);
}

std::unique_ptr<rtcp::RtcpPacket> RTCPSender::BuildPLI(const RtcpContext& ctx) {
  rtcp::Pli* pli = new rtcp::Pli();
  pli->SetSenderSsrc(ssrc_);
  pli->SetMediaSsrc(remote_ssrc_);

  TRACE_EVENT_INSTANT0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                       "RTCPSender::PLI");
  ++packet_type_counter_.pli_packets;
  TRACE_COUNTER_ID1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "RTCP_PLICount",
                    ssrc_, packet_type_counter_.pli_packets);

  return std::unique_ptr<rtcp::RtcpPacket>(pli);
}

std::unique_ptr<rtcp::RtcpPacket> RTCPSender::BuildFIR(const RtcpContext& ctx) {
  ++sequence_number_fir_;

  rtcp::Fir* fir = new rtcp::Fir();
  fir->SetSenderSsrc(ssrc_);
  fir->AddRequestTo(remote_ssrc_, sequence_number_fir_);

  TRACE_EVENT_INSTANT0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                       "RTCPSender::FIR");
  ++packet_type_counter_.fir_packets;
  TRACE_COUNTER_ID1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "RTCP_FIRCount",
                    ssrc_, packet_type_counter_.fir_packets);

  return std::unique_ptr<rtcp::RtcpPacket>(fir);
}

std::unique_ptr<rtcp::RtcpPacket> RTCPSender::BuildREMB(
    const RtcpContext& ctx) {
  rtcp::Remb* remb = new rtcp::Remb();
  remb->SetSenderSsrc(ssrc_);
  remb->SetBitrateBps(remb_bitrate_);
  remb->SetSsrcs(remb_ssrcs_);

  TRACE_EVENT_INSTANT0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                       "RTCPSender::REMB");

  return std::unique_ptr<rtcp::RtcpPacket>(remb);
}

void RTCPSender::SetTargetBitrate(unsigned int target_bitrate) {
  rtc::CritScope lock(&critical_section_rtcp_sender_);
  tmmbr_send_bps_ = target_bitrate;
}

std::unique_ptr<rtcp::RtcpPacket> RTCPSender::BuildTMMBR(
    const RtcpContext& ctx) {
  if (ctx.feedback_state_.module == nullptr)
    return nullptr;
  // 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 critical_section_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_.module->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 nullptr;
      }
    }
    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 nullptr;
      }
    }
  }

  if (!tmmbr_send_bps_)
    return nullptr;

  rtcp::Tmmbr* tmmbr = new rtcp::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);

  return std::unique_ptr<rtcp::RtcpPacket>(tmmbr);
}

std::unique_ptr<rtcp::RtcpPacket> RTCPSender::BuildTMMBN(
    const RtcpContext& ctx) {
  rtcp::Tmmbn* tmmbn = new rtcp::Tmmbn();
  tmmbn->SetSenderSsrc(ssrc_);
  for (const rtcp::TmmbItem& tmmbr : tmmbn_to_send_) {
    if (tmmbr.bitrate_bps() > 0) {
      tmmbn->AddTmmbr(tmmbr);
    }
  }

  return std::unique_ptr<rtcp::RtcpPacket>(tmmbn);
}

std::unique_ptr<rtcp::RtcpPacket> RTCPSender::BuildAPP(const RtcpContext& ctx) {
  rtcp::App* app = new rtcp::App();
  app->SetSsrc(ssrc_);
  app->SetSubType(app_sub_type_);
  app->SetName(app_name_);
  app->SetData(app_data_.get(), app_length_);

  return std::unique_ptr<rtcp::RtcpPacket>(app);
}

std::unique_ptr<rtcp::RtcpPacket> RTCPSender::BuildNACK(
    const RtcpContext& ctx) {
  rtcp::Nack* nack = new rtcp::Nack();
  nack->SetSenderSsrc(ssrc_);
  nack->SetMediaSsrc(remote_ssrc_);
  nack->SetPacketIds(ctx.nack_list_, ctx.nack_size_);

  // Report stats.
  NACKStringBuilder stringBuilder;
  for (int idx = 0; idx < ctx.nack_size_; ++idx) {
    stringBuilder.PushNACK(ctx.nack_list_[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();

  TRACE_EVENT_INSTANT1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                       "RTCPSender::NACK", "nacks",
                       TRACE_STR_COPY(stringBuilder.GetResult().c_str()));
  ++packet_type_counter_.nack_packets;
  TRACE_COUNTER_ID1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "RTCP_NACKCount",
                    ssrc_, packet_type_counter_.nack_packets);

  return std::unique_ptr<rtcp::RtcpPacket>(nack);
}

std::unique_ptr<rtcp::RtcpPacket> RTCPSender::BuildBYE(const RtcpContext& ctx) {
  rtcp::Bye* bye = new rtcp::Bye();
  bye->SetSenderSsrc(ssrc_);
  bye->SetCsrcs(csrcs_);

  return std::unique_ptr<rtcp::RtcpPacket>(bye);
}

std::unique_ptr<rtcp::RtcpPacket> RTCPSender::BuildExtendedReports(
    const RtcpContext& ctx) {
  std::unique_ptr<rtcp::ExtendedReports> xr(new rtcp::ExtendedReports());
  xr->SetSenderSsrc(ssrc_);

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

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

  if (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);
    video_bitrate_allocation_.reset();
  }

  if (xr_voip_metric_) {
    rtcp::VoipMetric voip;
    voip.SetMediaSsrc(remote_ssrc_);
    voip.SetVoipMetric(*xr_voip_metric_);
    xr_voip_metric_.reset();

    xr->SetVoipMetric(voip);
  }

  return std::move(xr);
}

int32_t RTCPSender::SendRTCP(const FeedbackState& feedback_state,
                             RTCPPacketType packetType,
                             int32_t nack_size,
                             const uint16_t* nack_list) {
  return SendCompoundRTCP(
      feedback_state, std::set<RTCPPacketType>(&packetType, &packetType + 1),
      nack_size, nack_list);
}

int32_t RTCPSender::SendCompoundRTCP(
    const FeedbackState& feedback_state,
    const std::set<RTCPPacketType>& packet_types,
    int32_t nack_size,
    const uint16_t* nack_list) {
  PacketContainer container(transport_, event_log_);
  size_t max_packet_size;

  {
    rtc::CritScope lock(&critical_section_rtcp_sender_);
    if (method_ == RtcpMode::kOff) {
      RTC_LOG(LS_WARNING) << "Can't send rtcp if it is disabled.";
      return -1;
    }
    // Add all flags as volatile. Non volatile entries will not be overwritten.
    // All new volatile flags added will be consumed by the end of this call.
    SetFlags(packet_types, true);

    // Prevent sending streams to send SR before any media has been sent.
    const bool can_calculate_rtp_timestamp = (last_frame_capture_time_ms_ >= 0);
    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_->CurrentNtpTime());

    PrepareReport(feedback_state);

    std::unique_ptr<rtcp::RtcpPacket> packet_bye;

    auto it = report_flags_.begin();
    while (it != report_flags_.end()) {
      auto builder_it = builders_.find(it->type);
      RTC_DCHECK(builder_it != builders_.end())
          << "Could not find builder for packet type " << it->type;
      if (it->is_volatile) {
        report_flags_.erase(it++);
      } else {
        ++it;
      }

      BuilderFunc func = builder_it->second;
      std::unique_ptr<rtcp::RtcpPacket> packet = (this->*func)(context);
      if (packet.get() == nullptr)
        return -1;
      // If there is a BYE, don't append now - save it and append it
      // at the end later.
      if (builder_it->first == kRtcpBye) {
        packet_bye = std::move(packet);
      } else {
        container.Append(packet.release());
      }
    }

    // Append the BYE now at the end
    if (packet_bye) {
      container.Append(packet_bye.release());
    }

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

    RTC_DCHECK(AllVolatileFlagsConsumed());
    max_packet_size = max_packet_size_;
  }

  size_t bytes_sent = container.SendPackets(max_packet_size);
  return bytes_sent == 0 ? -1 : 0;
}

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() || video_bitrate_allocation_) {
      SetFlag(kRtcpAnyExtendedReports, true);
    }

    // generate next time to send an RTCP report
    uint32_t minIntervalMs =
        rtc::dchecked_cast<uint32_t>(interval_config_.audio_interval_ms);

    if (!audio_) {
      if (sending_) {
        // Calculate bandwidth for video; 360 / send bandwidth in kbit/s.
        uint32_t send_bitrate_kbit = feedback_state.send_bitrate / 1000;
        if (send_bitrate_kbit != 0)
          minIntervalMs = 360000 / send_bitrate_kbit;
      }
      if (minIntervalMs >
          rtc::dchecked_cast<uint32_t>(interval_config_.video_interval_ms)) {
        minIntervalMs =
            rtc::dchecked_cast<uint32_t>(interval_config_.video_interval_ms);
      }
    }

    // The interval between RTCP packets is varied randomly over the
    // range [1/2,3/2] times the calculated interval.
    uint32_t timeToNext =
        random_.Rand(minIntervalMs * 1 / 2, minIntervalMs * 3 / 2);
    next_time_to_send_rtcp_ = clock_->TimeInMilliseconds() + timeToNext;

    // 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);
  rtc::CritScope lock(&critical_section_rtcp_sender_);
  csrcs_ = csrcs;
}

int32_t RTCPSender::SetApplicationSpecificData(uint8_t subType,
                                               uint32_t name,
                                               const uint8_t* data,
                                               uint16_t length) {
  if (length % 4 != 0) {
    RTC_LOG(LS_ERROR) << "Failed to SetApplicationSpecificData.";
    return -1;
  }
  rtc::CritScope lock(&critical_section_rtcp_sender_);

  SetFlag(kRtcpApp, true);
  app_sub_type_ = subType;
  app_name_ = name;
  app_data_.reset(new uint8_t[length]);
  app_length_ = length;
  memcpy(app_data_.get(), data, length);
  return 0;
}

// TODO(sprang): Remove support for VoIP metrics? (Not used in receiver.)
int32_t RTCPSender::SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric) {
  rtc::CritScope lock(&critical_section_rtcp_sender_);
  xr_voip_metric_.emplace(*VoIPMetric);

  SetFlag(kRtcpAnyExtendedReports, true);
  return 0;
}

void RTCPSender::SendRtcpXrReceiverReferenceTime(bool enable) {
  rtc::CritScope lock(&critical_section_rtcp_sender_);
  xr_send_receiver_reference_time_enabled_ = enable;
}

bool RTCPSender::RtcpXrReceiverReferenceTime() const {
  rtc::CritScope lock(&critical_section_rtcp_sender_);
  return xr_send_receiver_reference_time_enabled_;
}

void RTCPSender::SetTmmbn(std::vector<rtcp::TmmbItem> bounding_set) {
  rtc::CritScope lock(&critical_section_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));
  }
}

void RTCPSender::SetFlags(const std::set<RTCPPacketType>& types,
                          bool is_volatile) {
  for (RTCPPacketType type : types)
    SetFlag(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 BitrateAllocation& bitrate) {
  rtc::CritScope lock(&critical_section_rtcp_sender_);
  video_bitrate_allocation_.emplace(bitrate);
  SetFlag(kRtcpAnyExtendedReports, true);
}

bool RTCPSender::SendFeedbackPacket(const rtcp::TransportFeedback& packet) {
  size_t max_packet_size;
  {
    rtc::CritScope lock(&critical_section_rtcp_sender_);
    if (method_ == RtcpMode::kOff)
      return false;
    max_packet_size = max_packet_size_;
  }

  RTC_DCHECK_LE(max_packet_size, IP_PACKET_SIZE);
  bool send_failure = false;
  auto callback = [&](rtc::ArrayView<const uint8_t> packet) {
    if (transport_->SendRtcp(packet.data(), packet.size())) {
      if (event_log_)
        event_log_->Log(rtc::MakeUnique<RtcEventRtcpPacketOutgoing>(packet));
    } else {
      send_failure = true;
    }
  };
  return packet.Build(max_packet_size, callback) && !send_failure;
}

int64_t RTCPSender::RtcpAudioReportInverval() const {
  return interval_config_.audio_interval_ms;
}

int64_t RTCPSender::RtcpVideoReportInverval() const {
  return interval_config_.video_interval_ms;
}

}  // namespace webrtc