/*
 *  Copyright (c) 2024 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 "congestion_control_feedback_tracker.h"

#include <cstdint>
#include <tuple>
#include <vector>

#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "congestion_control_feedback.h"
#include "rtp_packet_received.h"

namespace webrtc {

void CongestionControlFeedbackTracker::ReceivedPacket(
    const RtpPacketReceived& packet) {
  int64_t unwrapped_sequence_number =
      unwrapper_.Unwrap(packet.SequenceNumber());
  if (last_sequence_number_in_feedback_ &&
      unwrapped_sequence_number < *last_sequence_number_in_feedback_ + 1) {
    LOG_WARN(
        "Received packet unorderered between feeedback. SSRC: {} Seq: {} last "
        "feedback: {}",
        packet.Ssrc(), packet.SequenceNumber(),
        static_cast<uint16_t>(*last_sequence_number_in_feedback_));
    // TODO: bugs.webrtc.org/374550342 - According to spec, the old packets
    // should be reported again. But at the moment, we dont store history of
    // packet we already reported and thus, they will be reported as lost. Note
    // that this is likely not a problem in webrtc since the packets will also
    // be removed from the send history when they are first reported as
    // received.
    last_sequence_number_in_feedback_ = unwrapped_sequence_number - 1;
  }
  packets_.push_back({packet.Ssrc(), unwrapped_sequence_number,
                      packet.arrival_time(), packet.ecn()});
}

void CongestionControlFeedbackTracker::AddPacketsToFeedback(
    Timestamp feedback_time,
    std::vector<rtcp::CongestionControlFeedback::PacketInfo>& packet_feedback) {
  if (packets_.empty()) {
    return;
  }
  std::sort(packets_.begin(), packets_.end(),
            [](const PacketInfo& a, const PacketInfo& b) {
              return std::tie(a.unwrapped_sequence_number, a.arrival_time) <
                     std::tie(b.unwrapped_sequence_number, b.arrival_time);
            });
  if (!last_sequence_number_in_feedback_) {
    last_sequence_number_in_feedback_ =
        packets_.front().unwrapped_sequence_number - 1;
  }

  auto packet_it = packets_.begin();
  uint32_t ssrc = packet_it->ssrc;
  for (int64_t sequence_number = *last_sequence_number_in_feedback_ + 1;
       sequence_number <= packets_.back().unwrapped_sequence_number;
       ++sequence_number) {
    rtc::EcnMarking ecn = rtc::EcnMarking::kNotEct;
    TimeDelta arrival_time_offset = TimeDelta::MinusInfinity();

    if (sequence_number == packet_it->unwrapped_sequence_number) {
      arrival_time_offset = feedback_time - packet_it->arrival_time;
      ecn = packet_it->ecn;
      ++packet_it;
      while (packet_it != packets_.end() &&
             packet_it->unwrapped_sequence_number == sequence_number) {
        // According to RFC 8888:
        // If duplicate copies of a particular RTP packet are received, then
        // the arrival time of the first copy to arrive MUST be reported. If
        // any of the copies of the duplicated packet are ECN-CE marked, then
        // an ECN-CE mark MUST be reported for that packet; otherwise, the ECN
        // mark of the first copy to arrive is reported.
        if (packet_it->ecn == rtc::EcnMarking::kCe) {
          ecn = rtc::EcnMarking::kCe;
        }
        LOG_WARN("Received duplicate packet ssrc:{} seq:{} ecn:{}", ssrc,
                 static_cast<uint16_t>(sequence_number), static_cast<int>(ecn));
        ++packet_it;
      }
    }  // else - the packet has not been received yet.
    packet_feedback.push_back({ssrc, static_cast<uint16_t>(sequence_number),
                               arrival_time_offset, ecn});
  }
  last_sequence_number_in_feedback_ = packets_.back().unwrapped_sequence_number;
  packets_.clear();
}

}  // namespace webrtc