text: /*
   *  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 "video/rtp_video_stream_receiver2.h"
  
  #include <algorithm>
  #include <limits>
  #include <memory>
  #include <utility>
  #include <vector>
  
  #include "absl/algorithm/container.h"
  #include "absl/base/macros.h"
  #include "absl/memory/memory.h"
  #include "absl/types/optional.h"
  #include "media/base/media_constants.h"
  #include "modules/pacing/packet_router.h"
  #include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
  #include "modules/rtp_rtcp/include/receive_statistics.h"
  #include "modules/rtp_rtcp/include/rtp_cvo.h"
  #include "modules/rtp_rtcp/include/ulpfec_receiver.h"
  #include "modules/rtp_rtcp/source/create_video_rtp_depacketizer.h"
  #include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
  #include "modules/rtp_rtcp/source/rtp_format.h"
  #include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h"
  #include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
  #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
  #include "modules/rtp_rtcp/source/rtp_packet_received.h"
  #include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
  #include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
  #include "modules/rtp_rtcp/source/video_rtp_depacketizer_raw.h"
  #include "modules/utility/include/process_thread.h"
  #include "modules/video_coding/frame_object.h"
  #include "modules/video_coding/h264_sprop_parameter_sets.h"
  #include "modules/video_coding/h264_sps_pps_tracker.h"
  #include "modules/video_coding/nack_module2.h"
  #include "modules/video_coding/packet_buffer.h"
  #include "rtc_base/checks.h"
  #include "rtc_base/location.h"
  #include "rtc_base/logging.h"
  #include "rtc_base/strings/string_builder.h"
  #include "system_wrappers/include/field_trial.h"
  #include "system_wrappers/include/metrics.h"
  #include "system_wrappers/include/ntp_time.h"
  #include "video/receive_statistics_proxy2.h"
  
  namespace webrtc {
  
  namespace {
  // TODO(philipel): Change kPacketBufferStartSize back to 32 in M63 see:
  //                 crbug.com/752886
  constexpr int kPacketBufferStartSize = 512;
  constexpr int kPacketBufferMaxSize = 2048;
  
  int PacketBufferMaxSize() {
    // The group here must be a positive power of 2, in which case that is used as
    // size. All other values shall result in the default value being used.
    const std::string group_name =
        webrtc::field_trial::FindFullName("WebRTC-PacketBufferMaxSize");
    int packet_buffer_max_size = kPacketBufferMaxSize;
    if (!group_name.empty() &&
        (sscanf(group_name.c_str(), "%d", &packet_buffer_max_size) != 1 ||
         packet_buffer_max_size <= 0 ||
         // Verify that the number is a positive power of 2.
         (packet_buffer_max_size & (packet_buffer_max_size - 1)) != 0)) {
      RTC_LOG(LS_WARNING) << "Invalid packet buffer max size: " << group_name;
      packet_buffer_max_size = kPacketBufferMaxSize;
    }
    return packet_buffer_max_size;
  }
  
  std::unique_ptr<ModuleRtpRtcpImpl2> CreateRtpRtcpModule(
      Clock* clock,
      ReceiveStatistics* receive_statistics,
      Transport* outgoing_transport,
      RtcpRttStats* rtt_stats,
      RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
      RtcpCnameCallback* rtcp_cname_callback,
      bool non_sender_rtt_measurement,
      uint32_t local_ssrc) {
    RtpRtcpInterface::Configuration configuration;
    configuration.clock = clock;
    configuration.audio = false;
    configuration.receiver_only = true;
    configuration.receive_statistics = receive_statistics;
    configuration.outgoing_transport = outgoing_transport;
    configuration.rtt_stats = rtt_stats;
    configuration.rtcp_packet_type_counter_observer =
        rtcp_packet_type_counter_observer;
    configuration.rtcp_cname_callback = rtcp_cname_callback;
    configuration.local_media_ssrc = local_ssrc;
    configuration.non_sender_rtt_measurement = non_sender_rtt_measurement;
  
    std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp =
        ModuleRtpRtcpImpl2::Create(configuration);
    rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
  
    return rtp_rtcp;
  }
  
  std::unique_ptr<NackModule2> MaybeConstructNackModule(
      TaskQueueBase* current_queue,
      const VideoReceiveStream::Config& config,
      Clock* clock,
      NackSender* nack_sender,
      KeyFrameRequestSender* keyframe_request_sender) {
    if (config.rtp.nack.rtp_history_ms == 0)
      return nullptr;
  
    return std::make_unique<NackModule2>(current_queue, clock, nack_sender,
                                         keyframe_request_sender);
  }
  
  static const int kPacketLogIntervalMs = 10000;
  
  }  // namespace
  
  RtpVideoStreamReceiver2::RtcpFeedbackBuffer::RtcpFeedbackBuffer(
      KeyFrameRequestSender* key_frame_request_sender,
      NackSender* nack_sender,
      LossNotificationSender* loss_notification_sender)
      : key_frame_request_sender_(key_frame_request_sender),
        nack_sender_(nack_sender),
        loss_notification_sender_(loss_notification_sender),
        request_key_frame_(false) {
    RTC_DCHECK(key_frame_request_sender_);
    RTC_DCHECK(nack_sender_);
    RTC_DCHECK(loss_notification_sender_);
  }
  
  void RtpVideoStreamReceiver2::RtcpFeedbackBuffer::RequestKeyFrame() {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
    request_key_frame_ = true;
  }
  
  void RtpVideoStreamReceiver2::RtcpFeedbackBuffer::SendNack(
      const std::vector<uint16_t>& sequence_numbers,
      bool buffering_allowed) {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
    RTC_DCHECK(!sequence_numbers.empty());
    nack_sequence_numbers_.insert(nack_sequence_numbers_.end(),
                                  sequence_numbers.cbegin(),
                                  sequence_numbers.cend());
    if (!buffering_allowed) {
      // Note that while *buffering* is not allowed, *batching* is, meaning that
      // previously buffered messages may be sent along with the current message.
      SendBufferedRtcpFeedback();
    }
  }
  
  void RtpVideoStreamReceiver2::RtcpFeedbackBuffer::SendLossNotification(
      uint16_t last_decoded_seq_num,
      uint16_t last_received_seq_num,
      bool decodability_flag,
      bool buffering_allowed) {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
    RTC_DCHECK(buffering_allowed);
    RTC_DCHECK(!lntf_state_)
        << "SendLossNotification() called twice in a row with no call to "
           "SendBufferedRtcpFeedback() in between.";
    lntf_state_ = absl::make_optional<LossNotificationState>(
        last_decoded_seq_num, last_received_seq_num, decodability_flag);
  }
  
  void RtpVideoStreamReceiver2::RtcpFeedbackBuffer::SendBufferedRtcpFeedback() {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
  
    bool request_key_frame = false;
    std::vector<uint16_t> nack_sequence_numbers;
    absl::optional<LossNotificationState> lntf_state;
  
    std::swap(request_key_frame, request_key_frame_);
    std::swap(nack_sequence_numbers, nack_sequence_numbers_);
    std::swap(lntf_state, lntf_state_);
  
    if (lntf_state) {
      // If either a NACK or a key frame request is sent, we should buffer
      // the LNTF and wait for them (NACK or key frame request) to trigger
      // the compound feedback message.
      // Otherwise, the LNTF should be sent out immediately.
      const bool buffering_allowed =
          request_key_frame || !nack_sequence_numbers.empty();
  
      loss_notification_sender_->SendLossNotification(
          lntf_state->last_decoded_seq_num, lntf_state->last_received_seq_num,
          lntf_state->decodability_flag, buffering_allowed);
    }
  
    if (request_key_frame) {
      key_frame_request_sender_->RequestKeyFrame();
    } else if (!nack_sequence_numbers.empty()) {
      nack_sender_->SendNack(nack_sequence_numbers, true);
    }
  }
  
  RtpVideoStreamReceiver2::RtpVideoStreamReceiver2(
      TaskQueueBase* current_queue,
      Clock* clock,
      Transport* transport,
      RtcpRttStats* rtt_stats,
      PacketRouter* packet_router,
      const VideoReceiveStream::Config* config,
      ReceiveStatistics* rtp_receive_statistics,
      RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
      RtcpCnameCallback* rtcp_cname_callback,
      ProcessThread* process_thread,
      NackSender* nack_sender,
      KeyFrameRequestSender* keyframe_request_sender,
      video_coding::OnCompleteFrameCallback* complete_frame_callback,
      rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
      rtc::scoped_refptr<FrameTransformerInterface> frame_transformer)
      : clock_(clock),
        config_(*config),
        packet_router_(packet_router),
        process_thread_(process_thread),
        ntp_estimator_(clock),
        rtp_header_extensions_(config_.rtp.extensions),
        forced_playout_delay_max_ms_("max_ms", absl::nullopt),
        forced_playout_delay_min_ms_("min_ms", absl::nullopt),
        rtp_receive_statistics_(rtp_receive_statistics),
        ulpfec_receiver_(UlpfecReceiver::Create(config->rtp.remote_ssrc,
                                                this,
                                                config->rtp.extensions)),
        receiving_(false),
        last_packet_log_ms_(-1),
        rtp_rtcp_(CreateRtpRtcpModule(
            clock,
            rtp_receive_statistics_,
            transport,
            rtt_stats,
            rtcp_packet_type_counter_observer,
            rtcp_cname_callback,
            config_.rtp.rtcp_xr.receiver_reference_time_report,
            config_.rtp.local_ssrc)),
        complete_frame_callback_(complete_frame_callback),
        keyframe_request_sender_(keyframe_request_sender),
        // TODO(bugs.webrtc.org/10336): Let |rtcp_feedback_buffer_| communicate
        // directly with |rtp_rtcp_|.
        rtcp_feedback_buffer_(this, nack_sender, this),
        nack_module_(MaybeConstructNackModule(current_queue,
                                              config_,
                                              clock_,
                                              &rtcp_feedback_buffer_,
                                              &rtcp_feedback_buffer_)),
        packet_buffer_(clock_, kPacketBufferStartSize, PacketBufferMaxSize()),
        has_received_frame_(false),
        frames_decryptable_(false),
        absolute_capture_time_receiver_(clock) {
    constexpr bool remb_candidate = true;
    if (packet_router_)
      packet_router_->AddReceiveRtpModule(rtp_rtcp_.get(), remb_candidate);
  
    RTC_DCHECK(config_.rtp.rtcp_mode != RtcpMode::kOff)
        << "A stream should not be configured with RTCP disabled. This value is "
           "reserved for internal usage.";
    // TODO(pbos): What's an appropriate local_ssrc for receive-only streams?
    RTC_DCHECK(config_.rtp.local_ssrc != 0);
    RTC_DCHECK(config_.rtp.remote_ssrc != config_.rtp.local_ssrc);
  
    rtp_rtcp_->SetRTCPStatus(config_.rtp.rtcp_mode);
    rtp_rtcp_->SetRemoteSSRC(config_.rtp.remote_ssrc);
  
    static const int kMaxPacketAgeToNack = 450;
    const int max_reordering_threshold = (config_.rtp.nack.rtp_history_ms > 0)
                                             ? kMaxPacketAgeToNack
                                             : kDefaultMaxReorderingThreshold;
    rtp_receive_statistics_->SetMaxReorderingThreshold(config_.rtp.remote_ssrc,
                                                       max_reordering_threshold);
    // TODO(nisse): For historic reasons, we applied the above
    // max_reordering_threshold also for RTX stats, which makes little sense since
    // we don't NACK rtx packets. Consider deleting the below block, and rely on
    // the default threshold.
    if (config_.rtp.rtx_ssrc) {
      rtp_receive_statistics_->SetMaxReorderingThreshold(
          config_.rtp.rtx_ssrc, max_reordering_threshold);
    }
  
    ParseFieldTrial(
        {&forced_playout_delay_max_ms_, &forced_playout_delay_min_ms_},
        field_trial::FindFullName("WebRTC-ForcePlayoutDelay"));
  
    process_thread_->RegisterModule(rtp_rtcp_.get(), RTC_FROM_HERE);
  
    if (config_.rtp.lntf.enabled) {
      loss_notification_controller_ =
          std::make_unique<LossNotificationController>(&rtcp_feedback_buffer_,
                                                       &rtcp_feedback_buffer_);
    }
  
    reference_finder_ =
        std::make_unique<video_coding::RtpFrameReferenceFinder>(this);
  
    // Only construct the encrypted receiver if frame encryption is enabled.
    if (config_.crypto_options.sframe.require_frame_encryption) {
      buffered_frame_decryptor_ =
          std::make_unique<BufferedFrameDecryptor>(this, this);
      if (frame_decryptor != nullptr) {
        buffered_frame_decryptor_->SetFrameDecryptor(std::move(frame_decryptor));
      }
    }
  
    if (frame_transformer) {
      frame_transformer_delegate_ = new rtc::RefCountedObject<
          RtpVideoStreamReceiverFrameTransformerDelegate>(
          this, std::move(frame_transformer), rtc::Thread::Current(),
          config_.rtp.remote_ssrc);
      frame_transformer_delegate_->Init();
    }
  }
  
  RtpVideoStreamReceiver2::~RtpVideoStreamReceiver2() {
    RTC_DCHECK(secondary_sinks_.empty());
  
    process_thread_->DeRegisterModule(rtp_rtcp_.get());
  
    if (packet_router_)
      packet_router_->RemoveReceiveRtpModule(rtp_rtcp_.get());
    UpdateHistograms();
    if (frame_transformer_delegate_)
      frame_transformer_delegate_->Reset();
  }
  
  void RtpVideoStreamReceiver2::AddReceiveCodec(
      uint8_t payload_type,
      const VideoCodec& video_codec,
      const std::map<std::string, std::string>& codec_params,
      bool raw_payload) {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
    if (codec_params.count(cricket::kH264FmtpSpsPpsIdrInKeyframe) ||
        field_trial::IsEnabled("WebRTC-SpsPpsIdrIsH264Keyframe")) {
      packet_buffer_.ForceSpsPpsIdrIsH264Keyframe();
    }
    payload_type_map_.emplace(
        payload_type, raw_payload
                          ? std::make_unique<VideoRtpDepacketizerRaw>()
                          : CreateVideoRtpDepacketizer(video_codec.codecType));
    pt_codec_params_.emplace(payload_type, codec_params);
  }
  
  absl::optional<Syncable::Info> RtpVideoStreamReceiver2::GetSyncInfo() const {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
    Syncable::Info info;
    if (rtp_rtcp_->RemoteNTP(&info.capture_time_ntp_secs,
                             &info.capture_time_ntp_frac, nullptr, nullptr,
                             &info.capture_time_source_clock) != 0) {
      return absl::nullopt;
    }
  
    if (!last_received_rtp_timestamp_ || !last_received_rtp_system_time_ms_) {
      return absl::nullopt;
    }
    info.latest_received_capture_timestamp = *last_received_rtp_timestamp_;
    info.latest_receive_time_ms = *last_received_rtp_system_time_ms_;
  
    // Leaves info.current_delay_ms uninitialized.
    return info;
  }
  
  RtpVideoStreamReceiver2::ParseGenericDependenciesResult
  RtpVideoStreamReceiver2::ParseGenericDependenciesExtension(
      const RtpPacketReceived& rtp_packet,
      RTPVideoHeader* video_header) {
    if (rtp_packet.HasExtension<RtpDependencyDescriptorExtension>()) {
      webrtc::DependencyDescriptor dependency_descriptor;
      if (!rtp_packet.GetExtension<RtpDependencyDescriptorExtension>(
              video_structure_.get(), &dependency_descriptor)) {
        // Descriptor is there, but failed to parse. Either it is invalid,
        // or too old packet (after relevant video_structure_ changed),
        // or too new packet (before relevant video_structure_ arrived).
        // Drop such packet to be on the safe side.
        // TODO(bugs.webrtc.org/10342): Stash too new packet.
        RTC_LOG(LS_WARNING) << "ssrc: " << rtp_packet.Ssrc()
                            << " Failed to parse dependency descriptor.";
        return kDropPacket;
      }
      if (dependency_descriptor.attached_structure != nullptr &&
          !dependency_descriptor.first_packet_in_frame) {
        RTC_LOG(LS_WARNING) << "ssrc: " << rtp_packet.Ssrc()
                            << "Invalid dependency descriptor: structure "
                               "attached to non first packet of a frame.";
        return kDropPacket;
      }
      video_header->is_first_packet_in_frame =
          dependency_descriptor.first_packet_in_frame;
      video_header->is_last_packet_in_frame =
          dependency_descriptor.last_packet_in_frame;
  
      int64_t frame_id =
          frame_id_unwrapper_.Unwrap(dependency_descriptor.frame_number);
      auto& generic_descriptor_info = video_header->generic.emplace();
      generic_descriptor_info.frame_id = frame_id;
      generic_descriptor_info.spatial_index =
          dependency_descriptor.frame_dependencies.spatial_id;
      generic_descriptor_info.temporal_index =
          dependency_descriptor.frame_dependencies.temporal_id;
      for (int fdiff : dependency_descriptor.frame_dependencies.frame_diffs) {
        generic_descriptor_info.dependencies.push_back(frame_id - fdiff);
      }
      generic_descriptor_info.decode_target_indications =
          dependency_descriptor.frame_dependencies.decode_target_indications;
      if (dependency_descriptor.resolution) {
        video_header->width = dependency_descriptor.resolution->Width();
        video_header->height = dependency_descriptor.resolution->Height();
      }
  
      // FrameDependencyStructure is sent in dependency descriptor of the first
      // packet of a key frame and required for parsed dependency descriptor in
      // all the following packets until next key frame.
      // Save it if there is a (potentially) new structure.
      if (dependency_descriptor.attached_structure) {
        RTC_DCHECK(dependency_descriptor.first_packet_in_frame);
        if (video_structure_frame_id_ > frame_id) {
          RTC_LOG(LS_WARNING)
              << "Arrived key frame with id " << frame_id << " and structure id "
              << dependency_descriptor.attached_structure->structure_id
              << " is older than the latest received key frame with id "
              << *video_structure_frame_id_ << " and structure id "
              << video_structure_->structure_id;
          return kDropPacket;
        }
        video_structure_ = std::move(dependency_descriptor.attached_structure);
        video_structure_frame_id_ = frame_id;
        video_header->frame_type = VideoFrameType::kVideoFrameKey;
      } else {
        video_header->frame_type = VideoFrameType::kVideoFrameDelta;
      }
      return kHasGenericDescriptor;
    }
  
    RtpGenericFrameDescriptor generic_frame_descriptor;
    if (!rtp_packet.GetExtension<RtpGenericFrameDescriptorExtension00>(
            &generic_frame_descriptor)) {
      return kNoGenericDescriptor;
    }
  
    video_header->is_first_packet_in_frame =
        generic_frame_descriptor.FirstPacketInSubFrame();
    video_header->is_last_packet_in_frame =
        generic_frame_descriptor.LastPacketInSubFrame();
  
    if (generic_frame_descriptor.FirstPacketInSubFrame()) {
      video_header->frame_type =
          generic_frame_descriptor.FrameDependenciesDiffs().empty()
              ? VideoFrameType::kVideoFrameKey
              : VideoFrameType::kVideoFrameDelta;
  
      auto& generic_descriptor_info = video_header->generic.emplace();
      int64_t frame_id =
          frame_id_unwrapper_.Unwrap(generic_frame_descriptor.FrameId());
      generic_descriptor_info.frame_id = frame_id;
      generic_descriptor_info.spatial_index =
          generic_frame_descriptor.SpatialLayer();
      generic_descriptor_info.temporal_index =
          generic_frame_descriptor.TemporalLayer();
      for (uint16_t fdiff : generic_frame_descriptor.FrameDependenciesDiffs()) {
        generic_descriptor_info.dependencies.push_back(frame_id - fdiff);
      }
    }
    video_header->width = generic_frame_descriptor.Width();
    video_header->height = generic_frame_descriptor.Height();
    return kHasGenericDescriptor;
  }
  
  void RtpVideoStreamReceiver2::OnReceivedPayloadData(
      rtc::CopyOnWriteBuffer codec_payload,
      const RtpPacketReceived& rtp_packet,
      const RTPVideoHeader& video) {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
    auto packet = std::make_unique<video_coding::PacketBuffer::Packet>(
        rtp_packet, video, ntp_estimator_.Estimate(rtp_packet.Timestamp()),
        clock_->TimeInMilliseconds());
  
    // Try to extrapolate absolute capture time if it is missing.
    packet->packet_info.set_absolute_capture_time(
        absolute_capture_time_receiver_.OnReceivePacket(
            AbsoluteCaptureTimeReceiver::GetSource(packet->packet_info.ssrc(),
                                                   packet->packet_info.csrcs()),
            packet->packet_info.rtp_timestamp(),
            // Assume frequency is the same one for all video frames.
            kVideoPayloadTypeFrequency,
            packet->packet_info.absolute_capture_time()));
  
    RTPVideoHeader& video_header = packet->video_header;
    video_header.rotation = kVideoRotation_0;
    video_header.content_type = VideoContentType::UNSPECIFIED;
    video_header.video_timing.flags = VideoSendTiming::kInvalid;
    video_header.is_last_packet_in_frame |= rtp_packet.Marker();
  
    if (const auto* vp9_header =
            absl::get_if<RTPVideoHeaderVP9>(&video_header.video_type_header)) {
      video_header.is_last_packet_in_frame |= vp9_header->end_of_frame;
      video_header.is_first_packet_in_frame |= vp9_header->beginning_of_frame;
    }
  
    rtp_packet.GetExtension<VideoOrientation>(&video_header.rotation);
    rtp_packet.GetExtension<VideoContentTypeExtension>(
        &video_header.content_type);
    rtp_packet.GetExtension<VideoTimingExtension>(&video_header.video_timing);
    if (forced_playout_delay_max_ms_ && forced_playout_delay_min_ms_) {
      video_header.playout_delay.max_ms = *forced_playout_delay_max_ms_;
      video_header.playout_delay.min_ms = *forced_playout_delay_min_ms_;
    } else {
      rtp_packet.GetExtension<PlayoutDelayLimits>(&video_header.playout_delay);
    }
  
    ParseGenericDependenciesResult generic_descriptor_state =
        ParseGenericDependenciesExtension(rtp_packet, &video_header);
    if (generic_descriptor_state == kDropPacket)
      return;
  
    // Color space should only be transmitted in the last packet of a frame,
    // therefore, neglect it otherwise so that last_color_space_ is not reset by
    // mistake.
    if (video_header.is_last_packet_in_frame) {
      video_header.color_space = rtp_packet.GetExtension<ColorSpaceExtension>();
      if (video_header.color_space ||
          video_header.frame_type == VideoFrameType::kVideoFrameKey) {
        // Store color space since it's only transmitted when changed or for key
        // frames. Color space will be cleared if a key frame is transmitted
        // without color space information.
        last_color_space_ = video_header.color_space;
      } else if (last_color_space_) {
        video_header.color_space = last_color_space_;
      }
    }
  
    if (loss_notification_controller_) {
      if (rtp_packet.recovered()) {
        // TODO(bugs.webrtc.org/10336): Implement support for reordering.
        RTC_LOG(LS_INFO)
            << "LossNotificationController does not support reordering.";
      } else if (generic_descriptor_state == kNoGenericDescriptor) {
        RTC_LOG(LS_WARNING) << "LossNotificationController requires generic "
                               "frame descriptor, but it is missing.";
      } else {
        if (video_header.is_first_packet_in_frame) {
          RTC_DCHECK(video_header.generic);
          LossNotificationController::FrameDetails frame;
          frame.is_keyframe =
              video_header.frame_type == VideoFrameType::kVideoFrameKey;
          frame.frame_id = video_header.generic->frame_id;
          frame.frame_dependencies = video_header.generic->dependencies;
          loss_notification_controller_->OnReceivedPacket(
              rtp_packet.SequenceNumber(), &frame);
        } else {
          loss_notification_controller_->OnReceivedPacket(
              rtp_packet.SequenceNumber(), nullptr);
        }
      }
    }
  
    if (nack_module_) {
      const bool is_keyframe =
          video_header.is_first_packet_in_frame &&
          video_header.frame_type == VideoFrameType::kVideoFrameKey;
  
      packet->times_nacked = nack_module_->OnReceivedPacket(
          rtp_packet.SequenceNumber(), is_keyframe, rtp_packet.recovered());
    } else {
      packet->times_nacked = -1;
    }
  
    if (codec_payload.size() == 0) {
      NotifyReceiverOfEmptyPacket(packet->seq_num);
      rtcp_feedback_buffer_.SendBufferedRtcpFeedback();
      return;
    }
  
    if (packet->codec() == kVideoCodecH264) {
      // Only when we start to receive packets will we know what payload type
      // that will be used. When we know the payload type insert the correct
      // sps/pps into the tracker.
      if (packet->payload_type != last_payload_type_) {
        last_payload_type_ = packet->payload_type;
        InsertSpsPpsIntoTracker(packet->payload_type);
      }
  
      video_coding::H264SpsPpsTracker::FixedBitstream fixed =
          tracker_.CopyAndFixBitstream(
              rtc::MakeArrayView(codec_payload.cdata(), codec_payload.size()),
              &packet->video_header);
  
      switch (fixed.action) {
        case video_coding::H264SpsPpsTracker::kRequestKeyframe:
          rtcp_feedback_buffer_.RequestKeyFrame();
          rtcp_feedback_buffer_.SendBufferedRtcpFeedback();
          ABSL_FALLTHROUGH_INTENDED;
        case video_coding::H264SpsPpsTracker::kDrop:
          return;
        case video_coding::H264SpsPpsTracker::kInsert:
          packet->video_payload = std::move(fixed.bitstream);
          break;
      }
  
    }
  #ifndef DISABLE_H265
    else if (packet->codec() == kVideoCodecH265) {
      // Only when we start to receive packets will we know what payload type
      // that will be used. When we know the payload type insert the correct
      // sps/pps into the tracker.
      if (packet->payload_type != last_payload_type_) {
        last_payload_type_ = packet->payload_type;
        InsertSpsPpsIntoTracker(packet->payload_type);
      }
  
      video_coding::H265VpsSpsPpsTracker::FixedBitstream fixed =
              h265_tracker_.CopyAndFixBitstream(
                      rtc::MakeArrayView(codec_payload.cdata(), codec_payload.size()),
                      &packet->video_header);
  
      switch (fixed.action) {
        case video_coding::H265VpsSpsPpsTracker::kRequestKeyframe:
          rtcp_feedback_buffer_.RequestKeyFrame();
              rtcp_feedback_buffer_.SendBufferedRtcpFeedback();
              ABSL_FALLTHROUGH_INTENDED;
        case video_coding::H265VpsSpsPpsTracker::kDrop:
          return;
        case video_coding::H265VpsSpsPpsTracker::kInsert:
          packet->video_payload = std::move(fixed.bitstream);
              break;
      }
    }
  #endif
    else {
      packet->video_payload = std::move(codec_payload);
    }
  
    rtcp_feedback_buffer_.SendBufferedRtcpFeedback();
    frame_counter_.Add(packet->timestamp);
    OnInsertedPacket(packet_buffer_.InsertPacket(std::move(packet)));
  }
  
  void RtpVideoStreamReceiver2::OnRecoveredPacket(const uint8_t* rtp_packet,
                                                  size_t rtp_packet_length) {
    RtpPacketReceived packet;
    if (!packet.Parse(rtp_packet, rtp_packet_length))
      return;
    if (packet.PayloadType() == config_.rtp.red_payload_type) {
      RTC_LOG(LS_WARNING) << "Discarding recovered packet with RED encapsulation";
      return;
    }
  
    packet.IdentifyExtensions(rtp_header_extensions_);
    packet.set_payload_type_frequency(kVideoPayloadTypeFrequency);
    // TODO(nisse): UlpfecReceiverImpl::ProcessReceivedFec passes both
    // original (decapsulated) media packets and recovered packets to
    // this callback. We need a way to distinguish, for setting
    // packet.recovered() correctly. Ideally, move RED decapsulation out
    // of the Ulpfec implementation.
  
    ReceivePacket(packet);
  }
  
  // This method handles both regular RTP packets and packets recovered
  // via FlexFEC.
  void RtpVideoStreamReceiver2::OnRtpPacket(const RtpPacketReceived& packet) {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
  
    if (!receiving_) {
      return;
    }
  
    if (!packet.recovered()) {
      // TODO(nisse): Exclude out-of-order packets?
      int64_t now_ms = clock_->TimeInMilliseconds();
  
      last_received_rtp_timestamp_ = packet.Timestamp();
      last_received_rtp_system_time_ms_ = now_ms;
  
      // Periodically log the RTP header of incoming packets.
      if (now_ms - last_packet_log_ms_ > kPacketLogIntervalMs) {
        rtc::StringBuilder ss;
        ss << "Packet received on SSRC: " << packet.Ssrc()
           << " with payload type: " << static_cast<int>(packet.PayloadType())
           << ", timestamp: " << packet.Timestamp()
           << ", sequence number: " << packet.SequenceNumber()
           << ", arrival time: " << packet.arrival_time_ms();
        int32_t time_offset;
        if (packet.GetExtension<TransmissionOffset>(&time_offset)) {
          ss << ", toffset: " << time_offset;
        }
        uint32_t send_time;
        if (packet.GetExtension<AbsoluteSendTime>(&send_time)) {
          ss << ", abs send time: " << send_time;
        }
        RTC_LOG(LS_INFO) << ss.str();
        last_packet_log_ms_ = now_ms;
      }
    }
  
    ReceivePacket(packet);
  
    // Update receive statistics after ReceivePacket.
    // Receive statistics will be reset if the payload type changes (make sure
    // that the first packet is included in the stats).
    if (!packet.recovered()) {
      rtp_receive_statistics_->OnRtpPacket(packet);
    }
  
    for (RtpPacketSinkInterface* secondary_sink : secondary_sinks_) {
      secondary_sink->OnRtpPacket(packet);
    }
  }
  
  void RtpVideoStreamReceiver2::RequestKeyFrame() {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
    // TODO(bugs.webrtc.org/10336): Allow the sender to ignore key frame requests
    // issued by anything other than the LossNotificationController if it (the
    // sender) is relying on LNTF alone.
    if (keyframe_request_sender_) {
      keyframe_request_sender_->RequestKeyFrame();
    } else {
      rtp_rtcp_->SendPictureLossIndication();
    }
  }
  
  void RtpVideoStreamReceiver2::SendLossNotification(
      uint16_t last_decoded_seq_num,
      uint16_t last_received_seq_num,
      bool decodability_flag,
      bool buffering_allowed) {
    RTC_DCHECK(config_.rtp.lntf.enabled);
    rtp_rtcp_->SendLossNotification(last_decoded_seq_num, last_received_seq_num,
                                    decodability_flag, buffering_allowed);
  }
  
  bool RtpVideoStreamReceiver2::IsUlpfecEnabled() const {
    return config_.rtp.ulpfec_payload_type != -1;
  }
  
  bool RtpVideoStreamReceiver2::IsRetransmissionsEnabled() const {
    return config_.rtp.nack.rtp_history_ms > 0;
  }
  
  void RtpVideoStreamReceiver2::RequestPacketRetransmit(
      const std::vector<uint16_t>& sequence_numbers) {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
    rtp_rtcp_->SendNack(sequence_numbers);
  }
  
  bool RtpVideoStreamReceiver2::IsDecryptable() const {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
    return frames_decryptable_;
  }
  
  void RtpVideoStreamReceiver2::OnInsertedPacket(
      video_coding::PacketBuffer::InsertResult result) {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
    video_coding::PacketBuffer::Packet* first_packet = nullptr;
    int max_nack_count;
    int64_t min_recv_time;
    int64_t max_recv_time;
    std::vector<rtc::ArrayView<const uint8_t>> payloads;
    RtpPacketInfos::vector_type packet_infos;
  
    bool frame_boundary = true;
    for (auto& packet : result.packets) {
      // PacketBuffer promisses frame boundaries are correctly set on each
      // packet. Document that assumption with the DCHECKs.
      RTC_DCHECK_EQ(frame_boundary, packet->is_first_packet_in_frame());
      if (packet->is_first_packet_in_frame()) {
        first_packet = packet.get();
        max_nack_count = packet->times_nacked;
        min_recv_time = packet->packet_info.receive_time_ms();
        max_recv_time = packet->packet_info.receive_time_ms();
        payloads.clear();
        packet_infos.clear();
      } else {
        max_nack_count = std::max(max_nack_count, packet->times_nacked);
        min_recv_time =
            std::min(min_recv_time, packet->packet_info.receive_time_ms());
        max_recv_time =
            std::max(max_recv_time, packet->packet_info.receive_time_ms());
      }
      payloads.emplace_back(packet->video_payload);
      packet_infos.push_back(packet->packet_info);
  
      frame_boundary = packet->is_last_packet_in_frame();
      if (packet->is_last_packet_in_frame()) {
        auto depacketizer_it = payload_type_map_.find(first_packet->payload_type);
        RTC_CHECK(depacketizer_it != payload_type_map_.end());
  
        rtc::scoped_refptr<EncodedImageBuffer> bitstream =
            depacketizer_it->second->AssembleFrame(payloads);
        if (!bitstream) {
          // Failed to assemble a frame. Discard and continue.
          continue;
        }
  
        const video_coding::PacketBuffer::Packet& last_packet = *packet;
        OnAssembledFrame(std::make_unique<video_coding::RtpFrameObject>(
            first_packet->seq_num,                    //
            last_packet.seq_num,                      //
            last_packet.marker_bit,                   //
            max_nack_count,                           //
            min_recv_time,                            //
            max_recv_time,                            //
            first_packet->timestamp,                  //
            first_packet->ntp_time_ms,                //
            last_packet.video_header.video_timing,    //
            first_packet->payload_type,               //
            first_packet->codec(),                    //
            last_packet.video_header.rotation,        //
            last_packet.video_header.content_type,    //
            first_packet->video_header,               //
            last_packet.video_header.color_space,     //
            RtpPacketInfos(std::move(packet_infos)),  //
            std::move(bitstream)));
      }
    }
    RTC_DCHECK(frame_boundary);
    if (result.buffer_cleared) {
      RequestKeyFrame();
    }
  }
  
  void RtpVideoStreamReceiver2::OnAssembledFrame(
      std::unique_ptr<video_coding::RtpFrameObject> frame) {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
    RTC_DCHECK(frame);
  
    const absl::optional<RTPVideoHeader::GenericDescriptorInfo>& descriptor =
        frame->GetRtpVideoHeader().generic;
  
    if (loss_notification_controller_ && descriptor) {
      loss_notification_controller_->OnAssembledFrame(
          frame->first_seq_num(), descriptor->frame_id,
          absl::c_linear_search(descriptor->decode_target_indications,
                                DecodeTargetIndication::kDiscardable),
          descriptor->dependencies);
    }
  
    // If frames arrive before a key frame, they would not be decodable.
    // In that case, request a key frame ASAP.
    if (!has_received_frame_) {
      if (frame->FrameType() != VideoFrameType::kVideoFrameKey) {
        // |loss_notification_controller_|, if present, would have already
        // requested a key frame when the first packet for the non-key frame
        // had arrived, so no need to replicate the request.
        if (!loss_notification_controller_) {
          RequestKeyFrame();
        }
      }
      has_received_frame_ = true;
    }
  
    // Reset |reference_finder_| if |frame| is new and the codec have changed.
    if (current_codec_) {
      bool frame_is_newer =
          AheadOf(frame->Timestamp(), last_assembled_frame_rtp_timestamp_);
  
      if (frame->codec_type() != current_codec_) {
        if (frame_is_newer) {
          // When we reset the |reference_finder_| we don't want new picture ids
          // to overlap with old picture ids. To ensure that doesn't happen we
          // start from the |last_completed_picture_id_| and add an offset in case
          // of reordering.
          reference_finder_ =
              std::make_unique<video_coding::RtpFrameReferenceFinder>(
                  this, last_completed_picture_id_ +
                            std::numeric_limits<uint16_t>::max());
          current_codec_ = frame->codec_type();
        } else {
          // Old frame from before the codec switch, discard it.
          return;
        }
      }
  
      if (frame_is_newer) {
        last_assembled_frame_rtp_timestamp_ = frame->Timestamp();
      }
    } else {
      current_codec_ = frame->codec_type();
      last_assembled_frame_rtp_timestamp_ = frame->Timestamp();
    }
  
    if (buffered_frame_decryptor_ != nullptr) {
      buffered_frame_decryptor_->ManageEncryptedFrame(std::move(frame));
    } else if (frame_transformer_delegate_) {
      frame_transformer_delegate_->TransformFrame(std::move(frame));
    } else {
      reference_finder_->ManageFrame(std::move(frame));
    }
  }
  
  void RtpVideoStreamReceiver2::OnCompleteFrame(
      std::unique_ptr<video_coding::EncodedFrame> frame) {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
    video_coding::RtpFrameObject* rtp_frame =
        static_cast<video_coding::RtpFrameObject*>(frame.get());
    last_seq_num_for_pic_id_[rtp_frame->id.picture_id] =
        rtp_frame->last_seq_num();
  
    last_completed_picture_id_ =
        std::max(last_completed_picture_id_, frame->id.picture_id);
    complete_frame_callback_->OnCompleteFrame(std::move(frame));
  }
  
  void RtpVideoStreamReceiver2::OnDecryptedFrame(
      std::unique_ptr<video_coding::RtpFrameObject> frame) {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
    reference_finder_->ManageFrame(std::move(frame));
  }
  
  void RtpVideoStreamReceiver2::OnDecryptionStatusChange(
      FrameDecryptorInterface::Status status) {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
    // Called from BufferedFrameDecryptor::DecryptFrame.
    frames_decryptable_ =
        (status == FrameDecryptorInterface::Status::kOk) ||
        (status == FrameDecryptorInterface::Status::kRecoverable);
  }
  
  void RtpVideoStreamReceiver2::SetFrameDecryptor(
      rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
    if (buffered_frame_decryptor_ == nullptr) {
      buffered_frame_decryptor_ =
          std::make_unique<BufferedFrameDecryptor>(this, this);
    }
    buffered_frame_decryptor_->SetFrameDecryptor(std::move(frame_decryptor));
  }
  
  void RtpVideoStreamReceiver2::SetDepacketizerToDecoderFrameTransformer(
      rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
    frame_transformer_delegate_ =
        new rtc::RefCountedObject<RtpVideoStreamReceiverFrameTransformerDelegate>(
            this, std::move(frame_transformer), rtc::Thread::Current(),
            config_.rtp.remote_ssrc);
    frame_transformer_delegate_->Init();
  }
  
  void RtpVideoStreamReceiver2::UpdateRtt(int64_t max_rtt_ms) {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
    if (nack_module_)
      nack_module_->UpdateRtt(max_rtt_ms);
  }
  
  absl::optional<int64_t> RtpVideoStreamReceiver2::LastReceivedPacketMs() const {
    return packet_buffer_.LastReceivedPacketMs();
  }
  
  absl::optional<int64_t> RtpVideoStreamReceiver2::LastReceivedKeyframePacketMs()
      const {
    return packet_buffer_.LastReceivedKeyframePacketMs();
  }
  
  void RtpVideoStreamReceiver2::AddSecondarySink(RtpPacketSinkInterface* sink) {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
    RTC_DCHECK(!absl::c_linear_search(secondary_sinks_, sink));
    secondary_sinks_.push_back(sink);
  }
  
  void RtpVideoStreamReceiver2::RemoveSecondarySink(
      const RtpPacketSinkInterface* sink) {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
    auto it = absl::c_find(secondary_sinks_, sink);
    if (it == secondary_sinks_.end()) {
      // We might be rolling-back a call whose setup failed mid-way. In such a
      // case, it's simpler to remove "everything" rather than remember what
      // has already been added.
      RTC_LOG(LS_WARNING) << "Removal of unknown sink.";
      return;
    }
    secondary_sinks_.erase(it);
  }
  
  void RtpVideoStreamReceiver2::ManageFrame(
      std::unique_ptr<video_coding::RtpFrameObject> frame) {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
    reference_finder_->ManageFrame(std::move(frame));
  }
  
  void RtpVideoStreamReceiver2::ReceivePacket(const RtpPacketReceived& packet) {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
    if (packet.payload_size() == 0) {
      // Padding or keep-alive packet.
      // TODO(nisse): Could drop empty packets earlier, but need to figure out how
      // they should be counted in stats.
      NotifyReceiverOfEmptyPacket(packet.SequenceNumber());
      return;
    }
    if (packet.PayloadType() == config_.rtp.red_payload_type) {
      ParseAndHandleEncapsulatingHeader(packet);
      return;
    }
  
    const auto type_it = payload_type_map_.find(packet.PayloadType());
    if (type_it == payload_type_map_.end()) {
      return;
    }
    absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed_payload =
        type_it->second->Parse(packet.PayloadBuffer());
    if (parsed_payload == absl::nullopt) {
      RTC_LOG(LS_WARNING) << "Failed parsing payload.";
      return;
    }
  
    OnReceivedPayloadData(std::move(parsed_payload->video_payload), packet,
                          parsed_payload->video_header);
  }
  
  void RtpVideoStreamReceiver2::ParseAndHandleEncapsulatingHeader(
      const RtpPacketReceived& packet) {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
    if (packet.PayloadType() == config_.rtp.red_payload_type &&
        packet.payload_size() > 0) {
      if (packet.payload()[0] == config_.rtp.ulpfec_payload_type) {
        // Notify video_receiver about received FEC packets to avoid NACKing these
        // packets.
        NotifyReceiverOfEmptyPacket(packet.SequenceNumber());
      }
      if (!ulpfec_receiver_->AddReceivedRedPacket(
              packet, config_.rtp.ulpfec_payload_type)) {
        return;
      }
      ulpfec_receiver_->ProcessReceivedFec();
    }
  }
  
  // In the case of a video stream without picture ids and no rtx the
  // RtpFrameReferenceFinder will need to know about padding to
  // correctly calculate frame references.
  void RtpVideoStreamReceiver2::NotifyReceiverOfEmptyPacket(uint16_t seq_num) {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
  
    reference_finder_->PaddingReceived(seq_num);
  
    OnInsertedPacket(packet_buffer_.InsertPadding(seq_num));
    if (nack_module_) {
      nack_module_->OnReceivedPacket(seq_num, /* is_keyframe = */ false,
                                     /* is _recovered = */ false);
    }
    if (loss_notification_controller_) {
      // TODO(bugs.webrtc.org/10336): Handle empty packets.
      RTC_LOG(LS_WARNING)
          << "LossNotificationController does not expect empty packets.";
    }
  }
  
  bool RtpVideoStreamReceiver2::DeliverRtcp(const uint8_t* rtcp_packet,
                                            size_t rtcp_packet_length) {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
  
    if (!receiving_) {
      return false;
    }
  
    rtp_rtcp_->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length);
  
    int64_t rtt = 0;
    rtp_rtcp_->RTT(config_.rtp.remote_ssrc, &rtt, nullptr, nullptr, nullptr);
    if (rtt == 0) {
      // Waiting for valid rtt.
      return true;
    }
    uint32_t ntp_secs = 0;
    uint32_t ntp_frac = 0;
    uint32_t rtp_timestamp = 0;
    uint32_t recieved_ntp_secs = 0;
    uint32_t recieved_ntp_frac = 0;
    if (rtp_rtcp_->RemoteNTP(&ntp_secs, &ntp_frac, &recieved_ntp_secs,
                             &recieved_ntp_frac, &rtp_timestamp) != 0) {
      // Waiting for RTCP.
      return true;
    }
    NtpTime recieved_ntp(recieved_ntp_secs, recieved_ntp_frac);
    int64_t time_since_recieved =
        clock_->CurrentNtpInMilliseconds() - recieved_ntp.ToMs();
    // Don't use old SRs to estimate time.
    if (time_since_recieved <= 1) {
      ntp_estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp);
      absl::optional<int64_t> remote_to_local_clock_offset_ms =
          ntp_estimator_.EstimateRemoteToLocalClockOffsetMs();
      if (remote_to_local_clock_offset_ms.has_value()) {
        absolute_capture_time_receiver_.SetRemoteToLocalClockOffset(
            Int64MsToQ32x32(*remote_to_local_clock_offset_ms));
      }
    }
  
    return true;
  }
  
  void RtpVideoStreamReceiver2::FrameContinuous(int64_t picture_id) {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
    if (!nack_module_)
      return;
  
    int seq_num = -1;
    auto seq_num_it = last_seq_num_for_pic_id_.find(picture_id);
    if (seq_num_it != last_seq_num_for_pic_id_.end())
      seq_num = seq_num_it->second;
    if (seq_num != -1)
      nack_module_->ClearUpTo(seq_num);
  }
  
  void RtpVideoStreamReceiver2::FrameDecoded(int64_t picture_id) {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
    // Running on the decoder thread.
    int seq_num = -1;
    auto seq_num_it = last_seq_num_for_pic_id_.find(picture_id);
    if (seq_num_it != last_seq_num_for_pic_id_.end()) {
      seq_num = seq_num_it->second;
      last_seq_num_for_pic_id_.erase(last_seq_num_for_pic_id_.begin(),
                                     ++seq_num_it);
    }
  
    if (seq_num != -1) {
      packet_buffer_.ClearTo(seq_num);
      reference_finder_->ClearTo(seq_num);
    }
  }
  
  void RtpVideoStreamReceiver2::SignalNetworkState(NetworkState state) {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
    rtp_rtcp_->SetRTCPStatus(state == kNetworkUp ? config_.rtp.rtcp_mode
                                                 : RtcpMode::kOff);
  }
  
  void RtpVideoStreamReceiver2::StartReceive() {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
    receiving_ = true;
  }
  
  void RtpVideoStreamReceiver2::StopReceive() {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
    receiving_ = false;
  }
  
  void RtpVideoStreamReceiver2::UpdateHistograms() {
    FecPacketCounter counter = ulpfec_receiver_->GetPacketCounter();
    if (counter.first_packet_time_ms == -1)
      return;
  
    int64_t elapsed_sec =
        (clock_->TimeInMilliseconds() - counter.first_packet_time_ms) / 1000;
    if (elapsed_sec < metrics::kMinRunTimeInSeconds)
      return;
  
    if (counter.num_packets > 0) {
      RTC_HISTOGRAM_PERCENTAGE(
          "WebRTC.Video.ReceivedFecPacketsInPercent",
          static_cast<int>(counter.num_fec_packets * 100 / counter.num_packets));
    }
    if (counter.num_fec_packets > 0) {
      RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.RecoveredMediaPacketsInPercentOfFec",
                               static_cast<int>(counter.num_recovered_packets *
                                                100 / counter.num_fec_packets));
    }
    if (config_.rtp.ulpfec_payload_type != -1) {
      RTC_HISTOGRAM_COUNTS_10000(
          "WebRTC.Video.FecBitrateReceivedInKbps",
          static_cast<int>(counter.num_bytes * 8 / elapsed_sec / 1000));
    }
  }
  
  void RtpVideoStreamReceiver2::InsertSpsPpsIntoTracker(uint8_t payload_type) {
    RTC_DCHECK_RUN_ON(&worker_task_checker_);
  
    auto codec_params_it = pt_codec_params_.find(payload_type);
    if (codec_params_it == pt_codec_params_.end())
      return;
  
    RTC_LOG(LS_INFO) << "Found out of band supplied codec parameters for"
                        " payload type: "
                     << static_cast<int>(payload_type);
  
    H264SpropParameterSets sprop_decoder;
    auto sprop_base64_it =
        codec_params_it->second.find(cricket::kH264FmtpSpropParameterSets);
  
    if (sprop_base64_it == codec_params_it->second.end())
      return;
  
    if (!sprop_decoder.DecodeSprop(sprop_base64_it->second.c_str()))
      return;
  
    tracker_.InsertSpsPpsNalus(sprop_decoder.sps_nalu(),
                               sprop_decoder.pps_nalu());
  }
  
  }  // namespace webrtc
  