/*
 *  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/bitrate_controller/send_side_bandwidth_estimation.h"

#include <algorithm>
#include <cmath>
#include <cstdio>
#include <limits>
#include <string>

#include "logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/ptr_util.h"
#include "system_wrappers/include/field_trial.h"
#include "system_wrappers/include/metrics.h"

namespace webrtc {
namespace {
const int64_t kBweIncreaseIntervalMs = 1000;
const int64_t kBweDecreaseIntervalMs = 300;
const int64_t kStartPhaseMs = 2000;
const int64_t kBweConverganceTimeMs = 20000;
const int kLimitNumPackets = 20;
const int kDefaultMaxBitrateBps = 1000000000;
const int64_t kLowBitrateLogPeriodMs = 10000;
const int64_t kRtcEventLogPeriodMs = 5000;
// Expecting that RTCP feedback is sent uniformly within [0.5, 1.5]s intervals.
const int64_t kFeedbackIntervalMs = 5000;
const int64_t kFeedbackTimeoutIntervals = 3;
const int64_t kTimeoutIntervalMs = 1000;

const float kDefaultLowLossThreshold = 0.02f;
const float kDefaultHighLossThreshold = 0.1f;
const int kDefaultBitrateThresholdKbps = 0;

struct UmaRampUpMetric {
  const char* metric_name;
  int bitrate_kbps;
};

const UmaRampUpMetric kUmaRampupMetrics[] = {
    {"WebRTC.BWE.RampUpTimeTo500kbpsInMs", 500},
    {"WebRTC.BWE.RampUpTimeTo1000kbpsInMs", 1000},
    {"WebRTC.BWE.RampUpTimeTo2000kbpsInMs", 2000}};
const size_t kNumUmaRampupMetrics =
    sizeof(kUmaRampupMetrics) / sizeof(kUmaRampupMetrics[0]);

const char kBweLosExperiment[] = "WebRTC-BweLossExperiment";

bool BweLossExperimentIsEnabled() {
  std::string experiment_string =
      webrtc::field_trial::FindFullName(kBweLosExperiment);
  // The experiment is enabled iff the field trial string begins with "Enabled".
  return experiment_string.find("Enabled") == 0;
}

bool ReadBweLossExperimentParameters(float* low_loss_threshold,
                                     float* high_loss_threshold,
                                     uint32_t* bitrate_threshold_kbps) {
  RTC_DCHECK(low_loss_threshold);
  RTC_DCHECK(high_loss_threshold);
  RTC_DCHECK(bitrate_threshold_kbps);
  std::string experiment_string =
      webrtc::field_trial::FindFullName(kBweLosExperiment);
  int parsed_values =
      sscanf(experiment_string.c_str(), "Enabled-%f,%f,%u", low_loss_threshold,
             high_loss_threshold, bitrate_threshold_kbps);
  if (parsed_values == 3) {
    RTC_CHECK_GT(*low_loss_threshold, 0.0f)
        << "Loss threshold must be greater than 0.";
    RTC_CHECK_LE(*low_loss_threshold, 1.0f)
        << "Loss threshold must be less than or equal to 1.";
    RTC_CHECK_GT(*high_loss_threshold, 0.0f)
        << "Loss threshold must be greater than 0.";
    RTC_CHECK_LE(*high_loss_threshold, 1.0f)
        << "Loss threshold must be less than or equal to 1.";
    RTC_CHECK_LE(*low_loss_threshold, *high_loss_threshold)
        << "The low loss threshold must be less than or equal to the high loss "
           "threshold.";
    RTC_CHECK_GE(*bitrate_threshold_kbps, 0)
        << "Bitrate threshold can't be negative.";
    RTC_CHECK_LT(*bitrate_threshold_kbps,
                 std::numeric_limits<int>::max() / 1000)
        << "Bitrate must be smaller enough to avoid overflows.";
    return true;
  }
  RTC_LOG(LS_WARNING) << "Failed to parse parameters for BweLossExperiment "
                         "experiment from field trial string. Using default.";
  *low_loss_threshold = kDefaultLowLossThreshold;
  *high_loss_threshold = kDefaultHighLossThreshold;
  *bitrate_threshold_kbps = kDefaultBitrateThresholdKbps;
  return false;
}
}  // namespace

SendSideBandwidthEstimation::SendSideBandwidthEstimation(RtcEventLog* event_log)
    : lost_packets_since_last_loss_update_Q8_(0),
      expected_packets_since_last_loss_update_(0),
      current_bitrate_bps_(0),
      min_bitrate_configured_(congestion_controller::GetMinBitrateBps()),
      max_bitrate_configured_(kDefaultMaxBitrateBps),
      last_low_bitrate_log_ms_(-1),
      has_decreased_since_last_fraction_loss_(false),
      last_feedback_ms_(-1),
      last_packet_report_ms_(-1),
      last_timeout_ms_(-1),
      last_fraction_loss_(0),
      last_logged_fraction_loss_(0),
      last_round_trip_time_ms_(0),
      bwe_incoming_(0),
      delay_based_bitrate_bps_(0),
      time_last_decrease_ms_(0),
      first_report_time_ms_(-1),
      initially_lost_packets_(0),
      bitrate_at_2_seconds_kbps_(0),
      uma_update_state_(kNoUpdate),
      rampup_uma_stats_updated_(kNumUmaRampupMetrics, false),
      event_log_(event_log),
      last_rtc_event_log_ms_(-1),
      in_timeout_experiment_(
          webrtc::field_trial::IsEnabled("WebRTC-FeedbackTimeout")),
      low_loss_threshold_(kDefaultLowLossThreshold),
      high_loss_threshold_(kDefaultHighLossThreshold),
      bitrate_threshold_bps_(1000 * kDefaultBitrateThresholdKbps) {
  RTC_DCHECK(event_log);
  if (BweLossExperimentIsEnabled()) {
    uint32_t bitrate_threshold_kbps;
    if (ReadBweLossExperimentParameters(&low_loss_threshold_,
                                        &high_loss_threshold_,
                                        &bitrate_threshold_kbps)) {
      RTC_LOG(LS_INFO) << "Enabled BweLossExperiment with parameters "
                       << low_loss_threshold_ << ", " << high_loss_threshold_
                       << ", " << bitrate_threshold_kbps;
      bitrate_threshold_bps_ = bitrate_threshold_kbps * 1000;
    }
  }
}

SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {}

void SendSideBandwidthEstimation::SetBitrates(int send_bitrate,
                                              int min_bitrate,
                                              int max_bitrate) {
  SetMinMaxBitrate(min_bitrate, max_bitrate);
  if (send_bitrate > 0)
    SetSendBitrate(send_bitrate);
}

void SendSideBandwidthEstimation::SetSendBitrate(int bitrate) {
  RTC_DCHECK_GT(bitrate, 0);
  delay_based_bitrate_bps_ = 0;  // Reset to avoid being capped by the estimate.
  CapBitrateToThresholds(Clock::GetRealTimeClock()->TimeInMilliseconds(),
                         bitrate);
  // Clear last sent bitrate history so the new value can be used directly
  // and not capped.
  min_bitrate_history_.clear();
}

void SendSideBandwidthEstimation::SetMinMaxBitrate(int min_bitrate,
                                                   int max_bitrate) {
  RTC_DCHECK_GE(min_bitrate, 0);
  min_bitrate_configured_ =
      std::max(min_bitrate, congestion_controller::GetMinBitrateBps());
  if (max_bitrate > 0) {
    max_bitrate_configured_ =
        std::max<uint32_t>(min_bitrate_configured_, max_bitrate);
  } else {
    max_bitrate_configured_ = kDefaultMaxBitrateBps;
  }
}

int SendSideBandwidthEstimation::GetMinBitrate() const {
  return min_bitrate_configured_;
}

void SendSideBandwidthEstimation::CurrentEstimate(int* bitrate,
                                                  uint8_t* loss,
                                                  int64_t* rtt) const {
  *bitrate = current_bitrate_bps_;
  *loss = last_fraction_loss_;
  *rtt = last_round_trip_time_ms_;
}

void SendSideBandwidthEstimation::UpdateReceiverEstimate(
    int64_t now_ms, uint32_t bandwidth) {
  bwe_incoming_ = bandwidth;
  CapBitrateToThresholds(now_ms, current_bitrate_bps_);
}

void SendSideBandwidthEstimation::UpdateDelayBasedEstimate(
    int64_t now_ms,
    uint32_t bitrate_bps) {
  delay_based_bitrate_bps_ = bitrate_bps;
  CapBitrateToThresholds(now_ms, current_bitrate_bps_);
}

void SendSideBandwidthEstimation::UpdateReceiverBlock(uint8_t fraction_loss,
                                                      int64_t rtt,
                                                      int number_of_packets,
                                                      int64_t now_ms) {
  last_feedback_ms_ = now_ms;
  if (first_report_time_ms_ == -1)
    first_report_time_ms_ = now_ms;

  // Update RTT if we were able to compute an RTT based on this RTCP.
  // FlexFEC doesn't send RTCP SR, which means we won't be able to compute RTT.
  if (rtt > 0)
    last_round_trip_time_ms_ = rtt;

  // Check sequence number diff and weight loss report
  if (number_of_packets > 0) {
    // Calculate number of lost packets.
    const int num_lost_packets_Q8 = fraction_loss * number_of_packets;
    // Accumulate reports.
    lost_packets_since_last_loss_update_Q8_ += num_lost_packets_Q8;
    expected_packets_since_last_loss_update_ += number_of_packets;

    // Don't generate a loss rate until it can be based on enough packets.
    if (expected_packets_since_last_loss_update_ < kLimitNumPackets)
      return;

    has_decreased_since_last_fraction_loss_ = false;
    last_fraction_loss_ = lost_packets_since_last_loss_update_Q8_ /
                          expected_packets_since_last_loss_update_;

    // Reset accumulators.
    lost_packets_since_last_loss_update_Q8_ = 0;
    expected_packets_since_last_loss_update_ = 0;
    last_packet_report_ms_ = now_ms;
    UpdateEstimate(now_ms);
  }
  UpdateUmaStats(now_ms, rtt, (fraction_loss * number_of_packets) >> 8);
}

void SendSideBandwidthEstimation::UpdateUmaStats(int64_t now_ms,
                                                 int64_t rtt,
                                                 int lost_packets) {
  int bitrate_kbps = static_cast<int>((current_bitrate_bps_ + 500) / 1000);
  for (size_t i = 0; i < kNumUmaRampupMetrics; ++i) {
    if (!rampup_uma_stats_updated_[i] &&
        bitrate_kbps >= kUmaRampupMetrics[i].bitrate_kbps) {
      RTC_HISTOGRAMS_COUNTS_100000(i, kUmaRampupMetrics[i].metric_name,
                                   now_ms - first_report_time_ms_);
      rampup_uma_stats_updated_[i] = true;
    }
  }
  if (IsInStartPhase(now_ms)) {
    initially_lost_packets_ += lost_packets;
  } else if (uma_update_state_ == kNoUpdate) {
    uma_update_state_ = kFirstDone;
    bitrate_at_2_seconds_kbps_ = bitrate_kbps;
    RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitiallyLostPackets",
                         initially_lost_packets_, 0, 100, 50);
    RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialRtt", static_cast<int>(rtt), 0,
                         2000, 50);
    RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialBandwidthEstimate",
                         bitrate_at_2_seconds_kbps_, 0, 2000, 50);
  } else if (uma_update_state_ == kFirstDone &&
             now_ms - first_report_time_ms_ >= kBweConverganceTimeMs) {
    uma_update_state_ = kDone;
    int bitrate_diff_kbps =
        std::max(bitrate_at_2_seconds_kbps_ - bitrate_kbps, 0);
    RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialVsConvergedDiff", bitrate_diff_kbps,
                         0, 2000, 50);
  }
}

void SendSideBandwidthEstimation::UpdateEstimate(int64_t now_ms) {
  uint32_t new_bitrate = current_bitrate_bps_;
  // We trust the REMB and/or delay-based estimate during the first 2 seconds if
  // we haven't had any packet loss reported, to allow startup bitrate probing.
  if (last_fraction_loss_ == 0 && IsInStartPhase(now_ms)) {
    new_bitrate = std::max(bwe_incoming_, new_bitrate);
    new_bitrate = std::max(delay_based_bitrate_bps_, new_bitrate);

    if (new_bitrate != current_bitrate_bps_) {
      min_bitrate_history_.clear();
      min_bitrate_history_.push_back(
          std::make_pair(now_ms, current_bitrate_bps_));
      CapBitrateToThresholds(now_ms, new_bitrate);
      return;
    }
  }
  UpdateMinHistory(now_ms);
  if (last_packet_report_ms_ == -1) {
    // No feedback received.
    CapBitrateToThresholds(now_ms, current_bitrate_bps_);
    return;
  }
  int64_t time_since_packet_report_ms = now_ms - last_packet_report_ms_;
  int64_t time_since_feedback_ms = now_ms - last_feedback_ms_;
  if (time_since_packet_report_ms < 1.2 * kFeedbackIntervalMs) {
    // We only care about loss above a given bitrate threshold.
    float loss = last_fraction_loss_ / 256.0f;
    // We only make decisions based on loss when the bitrate is above a
    // threshold. This is a crude way of handling loss which is uncorrelated
    // to congestion.
    if (current_bitrate_bps_ < bitrate_threshold_bps_ ||
        loss <= low_loss_threshold_) {
      // Loss < 2%: Increase rate by 8% of the min bitrate in the last
      // kBweIncreaseIntervalMs.
      // Note that by remembering the bitrate over the last second one can
      // rampup up one second faster than if only allowed to start ramping
      // at 8% per second rate now. E.g.:
      //   If sending a constant 100kbps it can rampup immediatly to 108kbps
      //   whenever a receiver report is received with lower packet loss.
      //   If instead one would do: current_bitrate_bps_ *= 1.08^(delta time),
      //   it would take over one second since the lower packet loss to achieve
      //   108kbps.
      new_bitrate = static_cast<uint32_t>(
          min_bitrate_history_.front().second * 1.08 + 0.5);

      // Add 1 kbps extra, just to make sure that we do not get stuck
      // (gives a little extra increase at low rates, negligible at higher
      // rates).
      new_bitrate += 1000;
    } else if (current_bitrate_bps_ > bitrate_threshold_bps_) {
      if (loss <= high_loss_threshold_) {
        // Loss between 2% - 10%: Do nothing.
      } else {
        // Loss > 10%: Limit the rate decreases to once a kBweDecreaseIntervalMs
        // + rtt.
        if (!has_decreased_since_last_fraction_loss_ &&
            (now_ms - time_last_decrease_ms_) >=
                (kBweDecreaseIntervalMs + last_round_trip_time_ms_)) {
          time_last_decrease_ms_ = now_ms;

          // Reduce rate:
          //   newRate = rate * (1 - 0.5*lossRate);
          //   where packetLoss = 256*lossRate;
          new_bitrate = static_cast<uint32_t>(
              (current_bitrate_bps_ *
               static_cast<double>(512 - last_fraction_loss_)) /
              512.0);
          has_decreased_since_last_fraction_loss_ = true;
        }
      }
    }
  } else if (time_since_feedback_ms >
                 kFeedbackTimeoutIntervals * kFeedbackIntervalMs &&
             (last_timeout_ms_ == -1 ||
              now_ms - last_timeout_ms_ > kTimeoutIntervalMs)) {
    if (in_timeout_experiment_) {
      RTC_LOG(LS_WARNING) << "Feedback timed out (" << time_since_feedback_ms
                          << " ms), reducing bitrate.";
      new_bitrate *= 0.8;
      // Reset accumulators since we've already acted on missing feedback and
      // shouldn't to act again on these old lost packets.
      lost_packets_since_last_loss_update_Q8_ = 0;
      expected_packets_since_last_loss_update_ = 0;
      last_timeout_ms_ = now_ms;
    }
  }

  CapBitrateToThresholds(now_ms, new_bitrate);
}

bool SendSideBandwidthEstimation::IsInStartPhase(int64_t now_ms) const {
  return first_report_time_ms_ == -1 ||
         now_ms - first_report_time_ms_ < kStartPhaseMs;
}

void SendSideBandwidthEstimation::UpdateMinHistory(int64_t now_ms) {
  // Remove old data points from history.
  // Since history precision is in ms, add one so it is able to increase
  // bitrate if it is off by as little as 0.5ms.
  while (!min_bitrate_history_.empty() &&
         now_ms - min_bitrate_history_.front().first + 1 >
             kBweIncreaseIntervalMs) {
    min_bitrate_history_.pop_front();
  }

  // Typical minimum sliding-window algorithm: Pop values higher than current
  // bitrate before pushing it.
  while (!min_bitrate_history_.empty() &&
         current_bitrate_bps_ <= min_bitrate_history_.back().second) {
    min_bitrate_history_.pop_back();
  }

  min_bitrate_history_.push_back(std::make_pair(now_ms, current_bitrate_bps_));
}

void SendSideBandwidthEstimation::CapBitrateToThresholds(int64_t now_ms,
                                                         uint32_t bitrate_bps) {
  if (bwe_incoming_ > 0 && bitrate_bps > bwe_incoming_) {
    bitrate_bps = bwe_incoming_;
  }
  if (delay_based_bitrate_bps_ > 0 && bitrate_bps > delay_based_bitrate_bps_) {
    bitrate_bps = delay_based_bitrate_bps_;
  }
  if (bitrate_bps > max_bitrate_configured_) {
    bitrate_bps = max_bitrate_configured_;
  }
  if (bitrate_bps < min_bitrate_configured_) {
    if (last_low_bitrate_log_ms_ == -1 ||
        now_ms - last_low_bitrate_log_ms_ > kLowBitrateLogPeriodMs) {
      RTC_LOG(LS_WARNING) << "Estimated available bandwidth "
                          << bitrate_bps / 1000
                          << " kbps is below configured min bitrate "
                          << min_bitrate_configured_ / 1000 << " kbps.";
      last_low_bitrate_log_ms_ = now_ms;
    }
    bitrate_bps = min_bitrate_configured_;
  }

  if (bitrate_bps != current_bitrate_bps_ ||
      last_fraction_loss_ != last_logged_fraction_loss_ ||
      now_ms - last_rtc_event_log_ms_ > kRtcEventLogPeriodMs) {
    event_log_->Log(rtc::MakeUnique<RtcEventBweUpdateLossBased>(
        bitrate_bps, last_fraction_loss_,
        expected_packets_since_last_loss_update_));
    last_logged_fraction_loss_ = last_fraction_loss_;
    last_rtc_event_log_ms_ = now_ms;
  }
  current_bitrate_bps_ = bitrate_bps;
}
}  // namespace webrtc