call/rtp_video_sender.cc

/*
 *  Copyright (c) 2015 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 "call/rtp_video_sender.h"

#include <algorithm>
#include <memory>
#include <string>
#include <utility>

#include "absl/memory/memory.h"
#include "api/transport/field_trial_based_config.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "modules/pacing/packet_router.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/playout_delay_oracle.h"
#include "modules/rtp_rtcp/source/rtp_sender.h"
#include "modules/utility/include/process_thread.h"
#include "modules/video_coding/include/video_codec_interface.h"
#include "rtc_base/checks.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "system_wrappers/include/field_trial.h"

namespace webrtc {

namespace webrtc_internal_rtp_video_sender {

RtpStreamSender::RtpStreamSender(
    std::unique_ptr<PlayoutDelayOracle> playout_delay_oracle,
    std::unique_ptr<RtpRtcp> rtp_rtcp,
    std::unique_ptr<RTPSenderVideo> sender_video)
    : playout_delay_oracle(std::move(playout_delay_oracle)),
      rtp_rtcp(std::move(rtp_rtcp)),
      sender_video(std::move(sender_video)) {}

RtpStreamSender::~RtpStreamSender() = default;

}  // namespace webrtc_internal_rtp_video_sender

namespace {
static const int kMinSendSidePacketHistorySize = 600;
// Assume an average video stream has around 3 packets per frame (1 mbps / 30
// fps / 1400B) A sequence number set with size 5500 will be able to store
// packet sequence number for at least last 60 seconds.
static const int kSendSideSeqNumSetMaxSize = 5500;
// We don't do MTU discovery, so assume that we have the standard ethernet MTU.
static const size_t kPathMTU = 1500;

using webrtc_internal_rtp_video_sender::RtpStreamSender;

std::vector<RtpStreamSender> CreateRtpStreamSenders(
    Clock* clock,
    const RtpConfig& rtp_config,
    int rtcp_report_interval_ms,
    Transport* send_transport,
    RtcpIntraFrameObserver* intra_frame_callback,
    RtcpBandwidthObserver* bandwidth_callback,
    RtpTransportControllerSendInterface* transport,
    RtcpRttStats* rtt_stats,
    FlexfecSender* flexfec_sender,
    BitrateStatisticsObserver* bitrate_observer,
    RtcpPacketTypeCounterObserver* rtcp_type_observer,
    SendSideDelayObserver* send_delay_observer,
    SendPacketObserver* send_packet_observer,
    RtcEventLog* event_log,
    RateLimiter* retransmission_rate_limiter,
    OverheadObserver* overhead_observer,
    RtpKeepAliveConfig keepalive_config,
    FrameEncryptorInterface* frame_encryptor,
    const CryptoOptions& crypto_options) {
  RTC_DCHECK_GT(rtp_config.ssrcs.size(), 0);

  RtpRtcp::Configuration configuration;
  configuration.clock = clock;
  configuration.audio = false;
  configuration.clock = Clock::GetRealTimeClock();
  configuration.receiver_only = false;
  configuration.outgoing_transport = send_transport;
  configuration.intra_frame_callback = intra_frame_callback;
  configuration.bandwidth_callback = bandwidth_callback;
  configuration.transport_feedback_callback =
      transport->transport_feedback_observer();
  configuration.rtt_stats = rtt_stats;
  configuration.rtcp_packet_type_counter_observer = rtcp_type_observer;
  configuration.paced_sender = transport->packet_sender();
  configuration.transport_sequence_number_allocator =
      transport->packet_router();
  configuration.send_bitrate_observer = bitrate_observer;
  configuration.send_side_delay_observer = send_delay_observer;
  configuration.send_packet_observer = send_packet_observer;
  configuration.event_log = event_log;
  configuration.retransmission_rate_limiter = retransmission_rate_limiter;
  configuration.overhead_observer = overhead_observer;
  configuration.keepalive_config = keepalive_config;
  configuration.frame_encryptor = frame_encryptor;
  configuration.require_frame_encryption =
      crypto_options.sframe.require_frame_encryption;
  configuration.extmap_allow_mixed = rtp_config.extmap_allow_mixed;
  configuration.rtcp_report_interval_ms = rtcp_report_interval_ms;

  std::vector<RtpStreamSender> rtp_streams;
  const std::vector<uint32_t>& flexfec_protected_ssrcs =
      rtp_config.flexfec.protected_media_ssrcs;
  for (uint32_t ssrc : rtp_config.ssrcs) {
    bool enable_flexfec = flexfec_sender != nullptr &&
                          std::find(flexfec_protected_ssrcs.begin(),
                                    flexfec_protected_ssrcs.end(),
                                    ssrc) != flexfec_protected_ssrcs.end();
    configuration.flexfec_sender = enable_flexfec ? flexfec_sender : nullptr;
    auto playout_delay_oracle = absl::make_unique<PlayoutDelayOracle>();

    configuration.ack_observer = playout_delay_oracle.get();
    auto rtp_rtcp = absl::WrapUnique(RtpRtcp::CreateRtpRtcp(configuration));
    rtp_rtcp->SetSendingStatus(false);
    rtp_rtcp->SetSendingMediaStatus(false);
    rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);

    auto sender_video = absl::make_unique<RTPSenderVideo>(
        configuration.clock, rtp_rtcp->RtpSender(), flexfec_sender,
        playout_delay_oracle.get(), frame_encryptor,
        crypto_options.sframe.require_frame_encryption,
        FieldTrialBasedConfig());
    rtp_streams.emplace_back(std::move(playout_delay_oracle),
                             std::move(rtp_rtcp), std::move(sender_video));
  }
  return rtp_streams;
}

bool PayloadTypeSupportsSkippingFecPackets(const std::string& payload_name) {
  const VideoCodecType codecType = PayloadStringToCodecType(payload_name);
  if (codecType == kVideoCodecVP8 || codecType == kVideoCodecVP9) {
    return true;
  }
  if (codecType == kVideoCodecGeneric &&
      field_trial::IsEnabled("WebRTC-GenericPictureId")) {
    return true;
  }
  return false;
}

// TODO(brandtr): Update this function when we support multistream protection.
std::unique_ptr<FlexfecSender> MaybeCreateFlexfecSender(
    Clock* clock,
    const RtpConfig& rtp,
    const std::map<uint32_t, RtpState>& suspended_ssrcs) {
  if (rtp.flexfec.payload_type < 0) {
    return nullptr;
  }
  RTC_DCHECK_GE(rtp.flexfec.payload_type, 0);
  RTC_DCHECK_LE(rtp.flexfec.payload_type, 127);
  if (rtp.flexfec.ssrc == 0) {
    RTC_LOG(LS_WARNING) << "FlexFEC is enabled, but no FlexFEC SSRC given. "
                           "Therefore disabling FlexFEC.";
    return nullptr;
  }
  if (rtp.flexfec.protected_media_ssrcs.empty()) {
    RTC_LOG(LS_WARNING)
        << "FlexFEC is enabled, but no protected media SSRC given. "
           "Therefore disabling FlexFEC.";
    return nullptr;
  }

  if (rtp.flexfec.protected_media_ssrcs.size() > 1) {
    RTC_LOG(LS_WARNING)
        << "The supplied FlexfecConfig contained multiple protected "
           "media streams, but our implementation currently only "
           "supports protecting a single media stream. "
           "To avoid confusion, disabling FlexFEC completely.";
    return nullptr;
  }

  const RtpState* rtp_state = nullptr;
  auto it = suspended_ssrcs.find(rtp.flexfec.ssrc);
  if (it != suspended_ssrcs.end()) {
    rtp_state = &it->second;
  }

  RTC_DCHECK_EQ(1U, rtp.flexfec.protected_media_ssrcs.size());
  return absl::make_unique<FlexfecSender>(
      rtp.flexfec.payload_type, rtp.flexfec.ssrc,
      rtp.flexfec.protected_media_ssrcs[0], rtp.mid, rtp.extensions,
      RTPSender::FecExtensionSizes(), rtp_state, clock);
}

uint32_t CalculateOverheadRateBps(int packets_per_second,
                                  size_t overhead_bytes_per_packet,
                                  uint32_t max_overhead_bps) {
  uint32_t overhead_bps =
      static_cast<uint32_t>(8 * overhead_bytes_per_packet * packets_per_second);
  return std::min(overhead_bps, max_overhead_bps);
}

int CalculatePacketRate(uint32_t bitrate_bps, size_t packet_size_bytes) {
  size_t packet_size_bits = 8 * packet_size_bytes;
  // Ceil for int value of bitrate_bps / packet_size_bits.
  return static_cast<int>((bitrate_bps + packet_size_bits - 1) /
                          packet_size_bits);
}
}  // namespace

RtpVideoSender::RtpVideoSender(
    Clock* clock,
    std::map<uint32_t, RtpState> suspended_ssrcs,
    const std::map<uint32_t, RtpPayloadState>& states,
    const RtpConfig& rtp_config,
    int rtcp_report_interval_ms,
    Transport* send_transport,
    const RtpSenderObservers& observers,
    RtpTransportControllerSendInterface* transport,
    RtcEventLog* event_log,
    RateLimiter* retransmission_limiter,
    std::unique_ptr<FecController> fec_controller,
    FrameEncryptorInterface* frame_encryptor,
    const CryptoOptions& crypto_options)
    : send_side_bwe_with_overhead_(
          webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
      account_for_packetization_overhead_(!webrtc::field_trial::IsDisabled(
          "WebRTC-SubtractPacketizationOverhead")),
      active_(false),
      module_process_thread_(nullptr),
      suspended_ssrcs_(std::move(suspended_ssrcs)),
      flexfec_sender_(
          MaybeCreateFlexfecSender(clock, rtp_config, suspended_ssrcs_)),
      fec_controller_(std::move(fec_controller)),
      rtp_streams_(CreateRtpStreamSenders(clock,
                                          rtp_config,
                                          rtcp_report_interval_ms,
                                          send_transport,
                                          observers.intra_frame_callback,
                                          transport->GetBandwidthObserver(),
                                          transport,
                                          observers.rtcp_rtt_stats,
                                          flexfec_sender_.get(),
                                          observers.bitrate_observer,
                                          observers.rtcp_type_observer,
                                          observers.send_delay_observer,
                                          observers.send_packet_observer,
                                          event_log,
                                          retransmission_limiter,
                                          this,
                                          // TODO(srte): Remove this argument.
                                          RtpKeepAliveConfig(),
                                          frame_encryptor,
                                          crypto_options)),
      rtp_config_(rtp_config),
      transport_(transport),
      transport_overhead_bytes_per_packet_(0),
      overhead_bytes_per_packet_(0),
      encoder_target_rate_bps_(0),
      frame_counts_(rtp_config.ssrcs.size()),
      frame_count_observer_(observers.frame_count_observer) {
  RTC_DCHECK_EQ(rtp_config.ssrcs.size(), rtp_streams_.size());
  module_process_thread_checker_.DetachFromThread();
  // SSRCs are assumed to be sorted in the same order as |rtp_modules|.
  for (uint32_t ssrc : rtp_config.ssrcs) {
    // Restore state if it previously existed.
    const RtpPayloadState* state = nullptr;
    auto it = states.find(ssrc);
    if (it != states.end()) {
      state = &it->second;
      shared_frame_id_ = std::max(shared_frame_id_, state->shared_frame_id);
    }
    params_.push_back(RtpPayloadParams(ssrc, state));
  }

  // RTP/RTCP initialization.

  // We add the highest spatial layer first to ensure it'll be prioritized
  // when sending padding, with the hope that the packet rate will be smaller,
  // and that it's more important to protect than the lower layers.

  // TODO(nisse): Consider moving registration with PacketRouter last, after the
  // modules are fully configured.
  for (const RtpStreamSender& stream : rtp_streams_) {
    constexpr bool remb_candidate = true;
    transport->packet_router()->AddSendRtpModule(stream.rtp_rtcp.get(),
                                                 remb_candidate);
  }

  for (size_t i = 0; i < rtp_config_.extensions.size(); ++i) {
    const std::string& extension = rtp_config_.extensions[i].uri;
    int id = rtp_config_.extensions[i].id;
    RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
    for (const RtpStreamSender& stream : rtp_streams_) {
      RTC_CHECK(stream.rtp_rtcp->RegisterRtpHeaderExtension(extension, id));
    }
  }

  ConfigureProtection(rtp_config);
  ConfigureSsrcs(rtp_config);
  ConfigureRids(rtp_config);

  if (!rtp_config.mid.empty()) {
    for (const RtpStreamSender& stream : rtp_streams_) {
      stream.rtp_rtcp->SetMid(rtp_config.mid);
    }
  }

  // TODO(pbos): Should we set CNAME on all RTP modules?
  rtp_streams_.front().rtp_rtcp->SetCNAME(rtp_config.c_name.c_str());

  for (const RtpStreamSender& stream : rtp_streams_) {
    stream.rtp_rtcp->RegisterRtcpStatisticsCallback(observers.rtcp_stats);
    stream.rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(
        observers.rtp_stats);
    stream.rtp_rtcp->SetMaxRtpPacketSize(rtp_config.max_packet_size);
    stream.rtp_rtcp->RegisterSendPayloadFrequency(rtp_config.payload_type,
                                                  kVideoPayloadTypeFrequency);
    stream.sender_video->RegisterPayloadType(rtp_config.payload_type,
                                             rtp_config.payload_name);
  }
  // Currently, both ULPFEC and FlexFEC use the same FEC rate calculation logic,
  // so enable that logic if either of those FEC schemes are enabled.
  fec_controller_->SetProtectionMethod(FecEnabled(), NackEnabled());

  fec_controller_->SetProtectionCallback(this);
  // Signal congestion controller this object is ready for OnPacket* callbacks.
  if (fec_controller_->UseLossVectorMask()) {
    transport_->RegisterPacketFeedbackObserver(this);
  }
}

RtpVideoSender::~RtpVideoSender() {
  for (const RtpStreamSender& stream : rtp_streams_) {
    transport_->packet_router()->RemoveSendRtpModule(stream.rtp_rtcp.get());
  }
  if (fec_controller_->UseLossVectorMask()) {
    transport_->DeRegisterPacketFeedbackObserver(this);
  }
}

void RtpVideoSender::RegisterProcessThread(
    ProcessThread* module_process_thread) {
  RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
  RTC_DCHECK(!module_process_thread_);
  module_process_thread_ = module_process_thread;

  for (const RtpStreamSender& stream : rtp_streams_) {
    module_process_thread_->RegisterModule(stream.rtp_rtcp.get(),
                                           RTC_FROM_HERE);
  }
}

void RtpVideoSender::DeRegisterProcessThread() {
  RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
  for (const RtpStreamSender& stream : rtp_streams_)
    module_process_thread_->DeRegisterModule(stream.rtp_rtcp.get());
}

void RtpVideoSender::SetActive(bool active) {
  rtc::CritScope lock(&crit_);
  if (active_ == active)
    return;
  const std::vector<bool> active_modules(rtp_streams_.size(), active);
  SetActiveModules(active_modules);
}

void RtpVideoSender::SetActiveModules(const std::vector<bool> active_modules) {
  rtc::CritScope lock(&crit_);
  RTC_DCHECK_EQ(rtp_streams_.size(), active_modules.size());
  active_ = false;
  for (size_t i = 0; i < active_modules.size(); ++i) {
    if (active_modules[i]) {
      active_ = true;
    }
    // Sends a kRtcpByeCode when going from true to false.
    rtp_streams_[i].rtp_rtcp->SetSendingStatus(active_modules[i]);
    // If set to false this module won't send media.
    rtp_streams_[i].rtp_rtcp->SetSendingMediaStatus(active_modules[i]);
  }
}

bool RtpVideoSender::IsActive() {
  rtc::CritScope lock(&crit_);
  return active_ && !rtp_streams_.empty();
}

EncodedImageCallback::Result RtpVideoSender::OnEncodedImage(
    const EncodedImage& encoded_image,
    const CodecSpecificInfo* codec_specific_info,
    const RTPFragmentationHeader* fragmentation) {
  fec_controller_->UpdateWithEncodedData(encoded_image.size(),
                                         encoded_image._frameType);
  rtc::CritScope lock(&crit_);
  RTC_DCHECK(!rtp_streams_.empty());
  if (!active_)
    return Result(Result::ERROR_SEND_FAILED);

  shared_frame_id_++;
  size_t stream_index = 0;
  if (codec_specific_info &&
      (codec_specific_info->codecType == kVideoCodecVP8 ||
       codec_specific_info->codecType == kVideoCodecH264 ||
       codec_specific_info->codecType == kVideoCodecGeneric)) {
    // Map spatial index to simulcast.
    stream_index = encoded_image.SpatialIndex().value_or(0);
  }
  RTC_DCHECK_LT(stream_index, rtp_streams_.size());
  RTPVideoHeader rtp_video_header = params_[stream_index].GetRtpVideoHeader(
      encoded_image, codec_specific_info, shared_frame_id_);

  uint32_t rtp_timestamp =
      encoded_image.Timestamp() +
      rtp_streams_[stream_index].rtp_rtcp->StartTimestamp();

  // RTCPSender has it's own copy of the timestamp offset, added in
  // RTCPSender::BuildSR, hence we must not add the in the offset for this call.
  // TODO(nisse): Delete RTCPSender:timestamp_offset_, and see if we can confine
  // knowledge of the offset to a single place.
  if (!rtp_streams_[stream_index].rtp_rtcp->OnSendingRtpFrame(
          encoded_image.Timestamp(), encoded_image.capture_time_ms_,
          rtp_config_.payload_type,
          encoded_image._frameType == kVideoFrameKey)) {
    // The payload router could be active but this module isn't sending.
    return Result(Result::ERROR_SEND_FAILED);
  }
  int64_t expected_retransmission_time_ms =
      rtp_streams_[stream_index].rtp_rtcp->ExpectedRetransmissionTimeMs();

  bool send_result = rtp_streams_[stream_index].sender_video->SendVideo(
      encoded_image._frameType, rtp_config_.payload_type, rtp_timestamp,
      encoded_image.capture_time_ms_, encoded_image.data(),
      encoded_image.size(), fragmentation, &rtp_video_header,
      expected_retransmission_time_ms);
  if (frame_count_observer_) {
    FrameCounts& counts = frame_counts_[stream_index];
    if (encoded_image._frameType == kVideoFrameKey) {
      ++counts.key_frames;
    } else if (encoded_image._frameType == kVideoFrameDelta) {
      ++counts.delta_frames;
    } else {
      RTC_DCHECK_EQ(encoded_image._frameType, kEmptyFrame);
    }
    frame_count_observer_->FrameCountUpdated(counts,
                                             rtp_config_.ssrcs[stream_index]);
  }
  if (!send_result)
    return Result(Result::ERROR_SEND_FAILED);

  return Result(Result::OK, rtp_timestamp);
}

void RtpVideoSender::OnBitrateAllocationUpdated(
    const VideoBitrateAllocation& bitrate) {
  rtc::CritScope lock(&crit_);
  if (IsActive()) {
    if (rtp_streams_.size() == 1) {
      // If spatial scalability is enabled, it is covered by a single stream.
      rtp_streams_[0].rtp_rtcp->SetVideoBitrateAllocation(bitrate);
    } else {
      std::vector<absl::optional<VideoBitrateAllocation>> layer_bitrates =
          bitrate.GetSimulcastAllocations();
      // Simulcast is in use, split the VideoBitrateAllocation into one struct
      // per rtp stream, moving over the temporal layer allocation.
      for (size_t i = 0; i < rtp_streams_.size(); ++i) {
        // The next spatial layer could be used if the current one is
        // inactive.
        if (layer_bitrates[i]) {
          rtp_streams_[i].rtp_rtcp->SetVideoBitrateAllocation(
              *layer_bitrates[i]);
        } else {
          // Signal a 0 bitrate on a simulcast stream.
          rtp_streams_[i].rtp_rtcp->SetVideoBitrateAllocation(
              VideoBitrateAllocation());
        }
      }
    }
  }
}

void RtpVideoSender::ConfigureProtection(const RtpConfig& rtp_config) {
  // Consistency of FlexFEC parameters is checked in MaybeCreateFlexfecSender.
  const bool flexfec_enabled = (flexfec_sender_ != nullptr);

  // Consistency of NACK and RED+ULPFEC parameters is checked in this function.
  const bool nack_enabled = rtp_config.nack.rtp_history_ms > 0;
  int red_payload_type = rtp_config.ulpfec.red_payload_type;
  int ulpfec_payload_type = rtp_config.ulpfec.ulpfec_payload_type;

  // Shorthands.
  auto IsRedEnabled = [&]() { return red_payload_type >= 0; };
  auto IsUlpfecEnabled = [&]() { return ulpfec_payload_type >= 0; };
  auto DisableRedAndUlpfec = [&]() {
    red_payload_type = -1;
    ulpfec_payload_type = -1;
  };

  if (webrtc::field_trial::IsEnabled("WebRTC-DisableUlpFecExperiment")) {
    RTC_LOG(LS_INFO) << "Experiment to disable sending ULPFEC is enabled.";
    DisableRedAndUlpfec();
  }

  // If enabled, FlexFEC takes priority over RED+ULPFEC.
  if (flexfec_enabled) {
    if (IsUlpfecEnabled()) {
      RTC_LOG(LS_INFO)
          << "Both FlexFEC and ULPFEC are configured. Disabling ULPFEC.";
    }
    DisableRedAndUlpfec();
  }

  // Payload types without picture ID cannot determine that a stream is complete
  // without retransmitting FEC, so using ULPFEC + NACK for H.264 (for instance)
  // is a waste of bandwidth since FEC packets still have to be transmitted.
  // Note that this is not the case with FlexFEC.
  if (nack_enabled && IsUlpfecEnabled() &&
      !PayloadTypeSupportsSkippingFecPackets(rtp_config.payload_name)) {
    RTC_LOG(LS_WARNING)
        << "Transmitting payload type without picture ID using "
           "NACK+ULPFEC is a waste of bandwidth since ULPFEC packets "
           "also have to be retransmitted. Disabling ULPFEC.";
    DisableRedAndUlpfec();
  }

  // Verify payload types.
  if (IsUlpfecEnabled() ^ IsRedEnabled()) {
    RTC_LOG(LS_WARNING)
        << "Only RED or only ULPFEC enabled, but not both. Disabling both.";
    DisableRedAndUlpfec();
  }

  for (const RtpStreamSender& stream : rtp_streams_) {
    // Set NACK.
    stream.rtp_rtcp->SetStorePacketsStatus(true, kMinSendSidePacketHistorySize);
    // Set RED/ULPFEC information.
    stream.sender_video->SetUlpfecConfig(red_payload_type, ulpfec_payload_type);
  }
}

bool RtpVideoSender::FecEnabled() const {
  const bool flexfec_enabled = (flexfec_sender_ != nullptr);
  const bool ulpfec_enabled =
      !webrtc::field_trial::IsEnabled("WebRTC-DisableUlpFecExperiment") &&
      (rtp_config_.ulpfec.ulpfec_payload_type >= 0);
  return flexfec_enabled || ulpfec_enabled;
}

bool RtpVideoSender::NackEnabled() const {
  const bool nack_enabled = rtp_config_.nack.rtp_history_ms > 0;
  return nack_enabled;
}

uint32_t RtpVideoSender::GetPacketizationOverheadRate() const {
  uint32_t packetization_overhead_bps = 0;
  for (size_t i = 0; i < rtp_streams_.size(); ++i) {
    if (rtp_streams_[i].rtp_rtcp->SendingMedia()) {
      packetization_overhead_bps +=
          rtp_streams_[i].sender_video->PacketizationOverheadBps();
    }
  }
  return packetization_overhead_bps;
}

void RtpVideoSender::DeliverRtcp(const uint8_t* packet, size_t length) {
  // Runs on a network thread.
  for (const RtpStreamSender& stream : rtp_streams_)
    stream.rtp_rtcp->IncomingRtcpPacket(packet, length);
}

void RtpVideoSender::ConfigureSsrcs(const RtpConfig& rtp_config) {
  // Configure regular SSRCs.
  for (size_t i = 0; i < rtp_config.ssrcs.size(); ++i) {
    uint32_t ssrc = rtp_config.ssrcs[i];
    RtpRtcp* const rtp_rtcp = rtp_streams_[i].rtp_rtcp.get();
    rtp_rtcp->SetSSRC(ssrc);

    // Restore RTP state if previous existed.
    auto it = suspended_ssrcs_.find(ssrc);
    if (it != suspended_ssrcs_.end())
      rtp_rtcp->SetRtpState(it->second);
  }

  // Set up RTX if available.
  if (rtp_config.rtx.ssrcs.empty())
    return;

  // Configure RTX SSRCs.
  RTC_DCHECK_EQ(rtp_config.rtx.ssrcs.size(), rtp_config.ssrcs.size());
  for (size_t i = 0; i < rtp_config.rtx.ssrcs.size(); ++i) {
    uint32_t ssrc = rtp_config.rtx.ssrcs[i];
    RtpRtcp* const rtp_rtcp = rtp_streams_[i].rtp_rtcp.get();
    rtp_rtcp->SetRtxSsrc(ssrc);
    auto it = suspended_ssrcs_.find(ssrc);
    if (it != suspended_ssrcs_.end())
      rtp_rtcp->SetRtxState(it->second);
  }

  // Configure RTX payload types.
  RTC_DCHECK_GE(rtp_config.rtx.payload_type, 0);
  for (const RtpStreamSender& stream : rtp_streams_) {
    stream.rtp_rtcp->SetRtxSendPayloadType(rtp_config.rtx.payload_type,
                                           rtp_config.payload_type);
    stream.rtp_rtcp->SetRtxSendStatus(kRtxRetransmitted |
                                      kRtxRedundantPayloads);
  }
  if (rtp_config.ulpfec.red_payload_type != -1 &&
      rtp_config.ulpfec.red_rtx_payload_type != -1) {
    for (const RtpStreamSender& stream : rtp_streams_) {
      stream.rtp_rtcp->SetRtxSendPayloadType(
          rtp_config.ulpfec.red_rtx_payload_type,
          rtp_config.ulpfec.red_payload_type);
    }
  }
}

void RtpVideoSender::ConfigureRids(const RtpConfig& rtp_config) {
  RTC_DCHECK(rtp_config.rids.empty() ||
             rtp_config.rids.size() == rtp_config.ssrcs.size());
  RTC_DCHECK(rtp_config.rids.empty() ||
             rtp_config.rids.size() == rtp_streams_.size());
  for (size_t i = 0; i < rtp_config.rids.size(); ++i) {
    const std::string& rid = rtp_config.rids[i];
    rtp_streams_[i].rtp_rtcp->SetRid(rid);
  }
}

void RtpVideoSender::OnNetworkAvailability(bool network_available) {
  for (const RtpStreamSender& stream : rtp_streams_) {
    stream.rtp_rtcp->SetRTCPStatus(network_available ? rtp_config_.rtcp_mode
                                                     : RtcpMode::kOff);
  }
}

std::map<uint32_t, RtpState> RtpVideoSender::GetRtpStates() const {
  std::map<uint32_t, RtpState> rtp_states;

  for (size_t i = 0; i < rtp_config_.ssrcs.size(); ++i) {
    uint32_t ssrc = rtp_config_.ssrcs[i];
    RTC_DCHECK_EQ(ssrc, rtp_streams_[i].rtp_rtcp->SSRC());
    rtp_states[ssrc] = rtp_streams_[i].rtp_rtcp->GetRtpState();
  }

  for (size_t i = 0; i < rtp_config_.rtx.ssrcs.size(); ++i) {
    uint32_t ssrc = rtp_config_.rtx.ssrcs[i];
    rtp_states[ssrc] = rtp_streams_[i].rtp_rtcp->GetRtxState();
  }

  if (flexfec_sender_) {
    uint32_t ssrc = rtp_config_.flexfec.ssrc;
    rtp_states[ssrc] = flexfec_sender_->GetRtpState();
  }

  return rtp_states;
}

std::map<uint32_t, RtpPayloadState> RtpVideoSender::GetRtpPayloadStates()
    const {
  rtc::CritScope lock(&crit_);
  std::map<uint32_t, RtpPayloadState> payload_states;
  for (const auto& param : params_) {
    payload_states[param.ssrc()] = param.state();
    payload_states[param.ssrc()].shared_frame_id = shared_frame_id_;
  }
  return payload_states;
}

void RtpVideoSender::OnTransportOverheadChanged(
    size_t transport_overhead_bytes_per_packet) {
  rtc::CritScope lock(&crit_);
  transport_overhead_bytes_per_packet_ = transport_overhead_bytes_per_packet;

  size_t max_rtp_packet_size =
      std::min(rtp_config_.max_packet_size,
               kPathMTU - transport_overhead_bytes_per_packet_);
  for (const RtpStreamSender& stream : rtp_streams_) {
    stream.rtp_rtcp->SetMaxRtpPacketSize(max_rtp_packet_size);
  }
}

void RtpVideoSender::OnOverheadChanged(size_t overhead_bytes_per_packet) {
  rtc::CritScope lock(&crit_);
  overhead_bytes_per_packet_ = overhead_bytes_per_packet;
}

void RtpVideoSender::OnBitrateUpdated(uint32_t bitrate_bps,
                                      uint8_t fraction_loss,
                                      int64_t rtt,
                                      int framerate) {
  // Substract overhead from bitrate.
  rtc::CritScope lock(&crit_);
  uint32_t payload_bitrate_bps = bitrate_bps;
  if (send_side_bwe_with_overhead_) {
    uint32_t overhead_bps = CalculateOverheadRateBps(
        CalculatePacketRate(
            bitrate_bps,
            rtp_config_.max_packet_size + transport_overhead_bytes_per_packet_),
        overhead_bytes_per_packet_ + transport_overhead_bytes_per_packet_,
        bitrate_bps);
    RTC_DCHECK_LE(overhead_bps, bitrate_bps);
    payload_bitrate_bps = bitrate_bps - overhead_bps;
  }

  // Get the encoder target rate. It is the estimated network rate -
  // protection overhead.
  encoder_target_rate_bps_ = fec_controller_->UpdateFecRates(
      payload_bitrate_bps, framerate, fraction_loss, loss_mask_vector_, rtt);

  uint32_t packetization_rate_bps = 0;
  if (account_for_packetization_overhead_) {
    // Subtract packetization overhead from the encoder target. If target rate
    // is really low, cap the overhead at 50%. This also avoids the case where
    // |encoder_target_rate_bps_| is 0 due to encoder pause event while the
    // packetization rate is positive since packets are still flowing.
    packetization_rate_bps =
        std::min(GetPacketizationOverheadRate(), encoder_target_rate_bps_ / 2);
    encoder_target_rate_bps_ -= packetization_rate_bps;
  }

  loss_mask_vector_.clear();

  uint32_t encoder_overhead_rate_bps =
      send_side_bwe_with_overhead_
          ? CalculateOverheadRateBps(
                CalculatePacketRate(encoder_target_rate_bps_,
                                    rtp_config_.max_packet_size +
                                        transport_overhead_bytes_per_packet_ -
                                        overhead_bytes_per_packet_),
                overhead_bytes_per_packet_ +
                    transport_overhead_bytes_per_packet_,
                bitrate_bps - encoder_target_rate_bps_)
          : 0;

  // When the field trial "WebRTC-SendSideBwe-WithOverhead" is enabled
  // protection_bitrate includes overhead.
  const uint32_t media_rate = encoder_target_rate_bps_ +
                              encoder_overhead_rate_bps +
                              packetization_rate_bps;
  RTC_DCHECK_GE(bitrate_bps, media_rate);
  protection_bitrate_bps_ = bitrate_bps - media_rate;
}

uint32_t RtpVideoSender::GetPayloadBitrateBps() const {
  return encoder_target_rate_bps_;
}

uint32_t RtpVideoSender::GetProtectionBitrateBps() const {
  return protection_bitrate_bps_;
}

int RtpVideoSender::ProtectionRequest(const FecProtectionParams* delta_params,
                                      const FecProtectionParams* key_params,
                                      uint32_t* sent_video_rate_bps,
                                      uint32_t* sent_nack_rate_bps,
                                      uint32_t* sent_fec_rate_bps) {
  *sent_video_rate_bps = 0;
  *sent_nack_rate_bps = 0;
  *sent_fec_rate_bps = 0;
  for (const RtpStreamSender& stream : rtp_streams_) {
    uint32_t not_used = 0;
    uint32_t module_nack_rate = 0;
    stream.sender_video->SetFecParameters(*delta_params, *key_params);
    *sent_video_rate_bps += stream.sender_video->VideoBitrateSent();
    *sent_fec_rate_bps += stream.sender_video->FecOverheadRate();
    stream.rtp_rtcp->BitrateSent(&not_used, /*video_rate=*/nullptr,
                                 /*fec_rate=*/nullptr, &module_nack_rate);
    *sent_nack_rate_bps += module_nack_rate;
  }
  return 0;
}

void RtpVideoSender::OnPacketAdded(uint32_t ssrc, uint16_t seq_num) {
  const auto ssrcs = rtp_config_.ssrcs;
  if (std::find(ssrcs.begin(), ssrcs.end(), ssrc) != ssrcs.end()) {
    feedback_packet_seq_num_set_.insert(seq_num);
    if (feedback_packet_seq_num_set_.size() > kSendSideSeqNumSetMaxSize) {
      RTC_LOG(LS_WARNING) << "Feedback packet sequence number set exceed it's "
                             "max size', will get reset.";
      feedback_packet_seq_num_set_.clear();
    }
  }
}

void RtpVideoSender::OnPacketFeedbackVector(
    const std::vector<PacketFeedback>& packet_feedback_vector) {
  rtc::CritScope lock(&crit_);
  // Lost feedbacks are not considered to be lost packets.
  for (const PacketFeedback& packet : packet_feedback_vector) {
    auto it = feedback_packet_seq_num_set_.find(packet.sequence_number);
    if (it != feedback_packet_seq_num_set_.end()) {
      const bool lost = packet.arrival_time_ms == PacketFeedback::kNotReceived;
      loss_mask_vector_.push_back(lost);
      feedback_packet_seq_num_set_.erase(it);
    }
  }
}

void RtpVideoSender::SetEncodingData(size_t width,
                                     size_t height,
                                     size_t num_temporal_layers) {
  fec_controller_->SetEncodingData(width, height, num_temporal_layers,
                                   rtp_config_.max_packet_size);
}
}  // namespace webrtc