/*
 *  Copyright (c) 2014 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/remote_bitrate_estimator/aimd_rate_control.h"

#include <inttypes.h>

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

#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "modules/remote_bitrate_estimator/overuse_detector.h"
#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_minmax.h"
#include "system_wrappers/include/field_trial.h"

namespace webrtc {

static const int64_t kDefaultRttMs = 200;
static const int64_t kMaxFeedbackIntervalMs = 1000;
static const float kDefaultBackoffFactor = 0.85f;
static const int64_t kDefaultInitialBackOffIntervalMs = 200;

const char kBweBackOffFactorExperiment[] = "WebRTC-BweBackOffFactor";
const char kBweInitialBackOffIntervalExperiment[] =
    "WebRTC-BweInitialBackOffInterval";

float ReadBackoffFactor() {
  std::string experiment_string =
      webrtc::field_trial::FindFullName(kBweBackOffFactorExperiment);
  float backoff_factor;
  int parsed_values =
      sscanf(experiment_string.c_str(), "Enabled-%f", &backoff_factor);
  if (parsed_values == 1) {
    if (backoff_factor >= 1.0f) {
      RTC_LOG(WARNING) << "Back-off factor must be less than 1.";
    } else if (backoff_factor <= 0.0f) {
      RTC_LOG(WARNING) << "Back-off factor must be greater than 0.";
    } else {
      return backoff_factor;
    }
  }
  RTC_LOG(LS_WARNING) << "Failed to parse parameters for AimdRateControl "
                         "experiment from field trial string. Using default.";
  return kDefaultBackoffFactor;
}

int64_t ReadInitialBackoffIntervalMs() {
  std::string experiment_string =
      webrtc::field_trial::FindFullName(kBweInitialBackOffIntervalExperiment);
  int64_t backoff_interval;
  int parsed_values =
      sscanf(experiment_string.c_str(), "Enabled-%" SCNd64, &backoff_interval);
  if (parsed_values == 1) {
    if (10 <= backoff_interval && backoff_interval <= 200) {
      return backoff_interval;
    }
    RTC_LOG(WARNING)
        << "Initial back-off interval must be between 10 and 200 ms.";
  }
  RTC_LOG(LS_WARNING) << "Failed to parse parameters for "
                      << kBweInitialBackOffIntervalExperiment
                      << " experiment. Using default.";
  return kDefaultInitialBackOffIntervalMs;
}

AimdRateControl::AimdRateControl()
    : min_configured_bitrate_bps_(congestion_controller::GetMinBitrateBps()),
      max_configured_bitrate_bps_(30000000),
      current_bitrate_bps_(max_configured_bitrate_bps_),
      latest_estimated_throughput_bps_(current_bitrate_bps_),
      avg_max_bitrate_kbps_(-1.0f),
      var_max_bitrate_kbps_(0.4f),
      rate_control_state_(kRcHold),
      rate_control_region_(kRcMaxUnknown),
      time_last_bitrate_change_(-1),
      time_last_bitrate_decrease_(-1),
      time_first_throughput_estimate_(-1),
      bitrate_is_initialized_(false),
      beta_(webrtc::field_trial::IsEnabled(kBweBackOffFactorExperiment)
                ? ReadBackoffFactor()
                : kDefaultBackoffFactor),
      rtt_(kDefaultRttMs),
      in_experiment_(!AdaptiveThresholdExperimentIsDisabled()),
      smoothing_experiment_(
          webrtc::field_trial::IsEnabled("WebRTC-Audio-BandwidthSmoothing")),
      in_initial_backoff_interval_experiment_(
          webrtc::field_trial::IsEnabled(kBweInitialBackOffIntervalExperiment)),
      initial_backoff_interval_ms_(kDefaultInitialBackOffIntervalMs) {
  if (in_initial_backoff_interval_experiment_) {
    initial_backoff_interval_ms_ = ReadInitialBackoffIntervalMs();
    RTC_LOG(LS_INFO) << "Using aimd rate control with initial back-off interval"
                     << " " << initial_backoff_interval_ms_ << " ms.";
  }
  RTC_LOG(LS_INFO) << "Using aimd rate control with back off factor " << beta_;
}

AimdRateControl::~AimdRateControl() {}

void AimdRateControl::SetStartBitrate(int start_bitrate_bps) {
  current_bitrate_bps_ = start_bitrate_bps;
  latest_estimated_throughput_bps_ = current_bitrate_bps_;
  bitrate_is_initialized_ = true;
}

void AimdRateControl::SetMinBitrate(int min_bitrate_bps) {
  min_configured_bitrate_bps_ = min_bitrate_bps;
  current_bitrate_bps_ = std::max<int>(min_bitrate_bps, current_bitrate_bps_);
}

bool AimdRateControl::ValidEstimate() const {
  return bitrate_is_initialized_;
}

int64_t AimdRateControl::GetFeedbackInterval() const {
  // Estimate how often we can send RTCP if we allocate up to 5% of bandwidth
  // to feedback.
  static const int kRtcpSize = 80;
  const int64_t interval = static_cast<int64_t>(
      kRtcpSize * 8.0 * 1000.0 / (0.05 * current_bitrate_bps_) + 0.5);
  const int64_t kMinFeedbackIntervalMs = 200;
  return rtc::SafeClamp(interval, kMinFeedbackIntervalMs,
                        kMaxFeedbackIntervalMs);
}

bool AimdRateControl::TimeToReduceFurther(
    int64_t now_ms,
    uint32_t estimated_throughput_bps) const {
  const int64_t bitrate_reduction_interval =
      std::max<int64_t>(std::min<int64_t>(rtt_, 200), 10);
  if (now_ms - time_last_bitrate_change_ >= bitrate_reduction_interval) {
    return true;
  }
  if (ValidEstimate()) {
    // TODO(terelius/holmer): Investigate consequences of increasing
    // the threshold to 0.95 * LatestEstimate().
    const uint32_t threshold = static_cast<uint32_t>(0.5 * LatestEstimate());
    return estimated_throughput_bps < threshold;
  }
  return false;
}

bool AimdRateControl::InitialTimeToReduceFurther(int64_t now_ms) const {
  if (!in_initial_backoff_interval_experiment_) {
    return ValidEstimate() &&
           TimeToReduceFurther(now_ms, LatestEstimate() / 2 - 1);
  }
  // TODO(terelius): We could use the RTT (clamped to suitable limits) instead
  // of a fixed bitrate_reduction_interval.
  if (time_last_bitrate_decrease_ == -1 ||
      now_ms - time_last_bitrate_decrease_ >= initial_backoff_interval_ms_) {
    return true;
  }
  return false;
}

uint32_t AimdRateControl::LatestEstimate() const {
  return current_bitrate_bps_;
}

void AimdRateControl::SetRtt(int64_t rtt) {
  rtt_ = rtt;
}

uint32_t AimdRateControl::Update(const RateControlInput* input,
                                 int64_t now_ms) {
  RTC_CHECK(input);

  // Set the initial bit rate value to what we're receiving the first half
  // second.
  // TODO(bugs.webrtc.org/9379): The comment above doesn't match to the code.
  if (!bitrate_is_initialized_) {
    const int64_t kInitializationTimeMs = 5000;
    RTC_DCHECK_LE(kBitrateWindowMs, kInitializationTimeMs);
    if (time_first_throughput_estimate_ < 0) {
      if (input->estimated_throughput_bps)
        time_first_throughput_estimate_ = now_ms;
    } else if (now_ms - time_first_throughput_estimate_ >
                   kInitializationTimeMs &&
               input->estimated_throughput_bps) {
      current_bitrate_bps_ = *input->estimated_throughput_bps;
      bitrate_is_initialized_ = true;
    }
  }

  current_bitrate_bps_ = ChangeBitrate(current_bitrate_bps_, *input, now_ms);
  return current_bitrate_bps_;
}

void AimdRateControl::SetEstimate(int bitrate_bps, int64_t now_ms) {
  bitrate_is_initialized_ = true;
  uint32_t prev_bitrate_bps = current_bitrate_bps_;
  current_bitrate_bps_ = ClampBitrate(bitrate_bps, bitrate_bps);
  time_last_bitrate_change_ = now_ms;
  if (current_bitrate_bps_ < prev_bitrate_bps) {
    time_last_bitrate_decrease_ = now_ms;
  }
}

int AimdRateControl::GetNearMaxIncreaseRateBps() const {
  RTC_DCHECK_GT(current_bitrate_bps_, 0);
  double bits_per_frame = static_cast<double>(current_bitrate_bps_) / 30.0;
  double packets_per_frame = std::ceil(bits_per_frame / (8.0 * 1200.0));
  double avg_packet_size_bits = bits_per_frame / packets_per_frame;

  // Approximate the over-use estimator delay to 100 ms.
  const int64_t response_time = in_experiment_ ? (rtt_ + 100) * 2 : rtt_ + 100;
  constexpr double kMinIncreaseRateBps = 4000;
  return static_cast<int>(std::max(
      kMinIncreaseRateBps, (avg_packet_size_bits * 1000) / response_time));
}

int AimdRateControl::GetExpectedBandwidthPeriodMs() const {
  const int kMinPeriodMs = smoothing_experiment_ ? 500 : 2000;
  constexpr int kDefaultPeriodMs = 3000;
  constexpr int kMaxPeriodMs = 50000;

  int increase_rate = GetNearMaxIncreaseRateBps();
  if (!last_decrease_)
    return smoothing_experiment_ ? kMinPeriodMs : kDefaultPeriodMs;

  return std::min(kMaxPeriodMs,
                  std::max<int>(1000 * static_cast<int64_t>(*last_decrease_) /
                                    increase_rate,
                                kMinPeriodMs));
}

uint32_t AimdRateControl::ChangeBitrate(uint32_t new_bitrate_bps,
                                        const RateControlInput& input,
                                        int64_t now_ms) {
  uint32_t estimated_throughput_bps =
      input.estimated_throughput_bps.value_or(latest_estimated_throughput_bps_);
  if (input.estimated_throughput_bps)
    latest_estimated_throughput_bps_ = *input.estimated_throughput_bps;

  // An over-use should always trigger us to reduce the bitrate, even though
  // we have not yet established our first estimate. By acting on the over-use,
  // we will end up with a valid estimate.
  if (!bitrate_is_initialized_ &&
      input.bw_state != BandwidthUsage::kBwOverusing)
    return current_bitrate_bps_;

  ChangeState(input, now_ms);
  // Calculated here because it's used in multiple places.
  const float estimated_throughput_kbps = estimated_throughput_bps / 1000.0f;
  // Calculate the max bit rate std dev given the normalized
  // variance and the current throughput bitrate.
  const float std_max_bit_rate =
      sqrt(var_max_bitrate_kbps_ * avg_max_bitrate_kbps_);
  switch (rate_control_state_) {
    case kRcHold:
      break;

    case kRcIncrease:
      if (avg_max_bitrate_kbps_ >= 0 &&
          estimated_throughput_kbps >
              avg_max_bitrate_kbps_ + 3 * std_max_bit_rate) {
        rate_control_region_ = kRcMaxUnknown;

        avg_max_bitrate_kbps_ = -1.0;
      }
      if (rate_control_region_ == kRcNearMax) {
        uint32_t additive_increase_bps =
            AdditiveRateIncrease(now_ms, time_last_bitrate_change_);
        new_bitrate_bps += additive_increase_bps;
      } else {
        uint32_t multiplicative_increase_bps = MultiplicativeRateIncrease(
            now_ms, time_last_bitrate_change_, new_bitrate_bps);
        new_bitrate_bps += multiplicative_increase_bps;
      }

      time_last_bitrate_change_ = now_ms;
      break;

    case kRcDecrease:
      // Set bit rate to something slightly lower than max
      // to get rid of any self-induced delay.
      new_bitrate_bps =
          static_cast<uint32_t>(beta_ * estimated_throughput_bps + 0.5);
      if (new_bitrate_bps > current_bitrate_bps_) {
        // Avoid increasing the rate when over-using.
        if (rate_control_region_ != kRcMaxUnknown) {
          new_bitrate_bps = static_cast<uint32_t>(
              beta_ * avg_max_bitrate_kbps_ * 1000 + 0.5f);
        }
        new_bitrate_bps = std::min(new_bitrate_bps, current_bitrate_bps_);
      }
      rate_control_region_ = kRcNearMax;

      if (bitrate_is_initialized_ &&
          estimated_throughput_bps < current_bitrate_bps_) {
        constexpr float kDegradationFactor = 0.9f;
        if (smoothing_experiment_ &&
            new_bitrate_bps <
                kDegradationFactor * beta_ * current_bitrate_bps_) {
          // If bitrate decreases more than a normal back off after overuse, it
          // indicates a real network degradation. We do not let such a decrease
          // to determine the bandwidth estimation period.
          last_decrease_ = absl::nullopt;
        } else {
          last_decrease_ = current_bitrate_bps_ - new_bitrate_bps;
        }
      }
      if (estimated_throughput_kbps <
          avg_max_bitrate_kbps_ - 3 * std_max_bit_rate) {
        avg_max_bitrate_kbps_ = -1.0f;
      }

      bitrate_is_initialized_ = true;
      UpdateMaxThroughputEstimate(estimated_throughput_kbps);
      // Stay on hold until the pipes are cleared.
      rate_control_state_ = kRcHold;
      time_last_bitrate_change_ = now_ms;
      time_last_bitrate_decrease_ = now_ms;
      break;

    default:
      assert(false);
  }
  return ClampBitrate(new_bitrate_bps, estimated_throughput_bps);
}

uint32_t AimdRateControl::ClampBitrate(
    uint32_t new_bitrate_bps,
    uint32_t estimated_throughput_bps) const {
  // Don't change the bit rate if the send side is too far off.
  // We allow a bit more lag at very low rates to not too easily get stuck if
  // the encoder produces uneven outputs.
  const uint32_t max_bitrate_bps =
      static_cast<uint32_t>(1.5f * estimated_throughput_bps) + 10000;
  if (new_bitrate_bps > current_bitrate_bps_ &&
      new_bitrate_bps > max_bitrate_bps) {
    new_bitrate_bps = std::max(current_bitrate_bps_, max_bitrate_bps);
  }
  new_bitrate_bps = std::max(new_bitrate_bps, min_configured_bitrate_bps_);
  return new_bitrate_bps;
}

uint32_t AimdRateControl::MultiplicativeRateIncrease(
    int64_t now_ms,
    int64_t last_ms,
    uint32_t current_bitrate_bps) const {
  double alpha = 1.08;
  if (last_ms > -1) {
    auto time_since_last_update_ms =
        rtc::SafeMin<int64_t>(now_ms - last_ms, 1000);
    alpha = pow(alpha, time_since_last_update_ms / 1000.0);
  }
  uint32_t multiplicative_increase_bps =
      std::max(current_bitrate_bps * (alpha - 1.0), 1000.0);
  return multiplicative_increase_bps;
}

uint32_t AimdRateControl::AdditiveRateIncrease(int64_t now_ms,
                                               int64_t last_ms) const {
  return static_cast<uint32_t>((now_ms - last_ms) *
                               GetNearMaxIncreaseRateBps() / 1000);
}

void AimdRateControl::UpdateMaxThroughputEstimate(
    float estimated_throughput_kbps) {
  const float alpha = 0.05f;
  if (avg_max_bitrate_kbps_ == -1.0f) {
    avg_max_bitrate_kbps_ = estimated_throughput_kbps;
  } else {
    avg_max_bitrate_kbps_ =
        (1 - alpha) * avg_max_bitrate_kbps_ + alpha * estimated_throughput_kbps;
  }
  // Estimate the max bit rate variance and normalize the variance
  // with the average max bit rate.
  const float norm = std::max(avg_max_bitrate_kbps_, 1.0f);
  var_max_bitrate_kbps_ =
      (1 - alpha) * var_max_bitrate_kbps_ +
      alpha * (avg_max_bitrate_kbps_ - estimated_throughput_kbps) *
          (avg_max_bitrate_kbps_ - estimated_throughput_kbps) / norm;
  // 0.4 ~= 14 kbit/s at 500 kbit/s
  if (var_max_bitrate_kbps_ < 0.4f) {
    var_max_bitrate_kbps_ = 0.4f;
  }
  // 2.5f ~= 35 kbit/s at 500 kbit/s
  if (var_max_bitrate_kbps_ > 2.5f) {
    var_max_bitrate_kbps_ = 2.5f;
  }
}

void AimdRateControl::ChangeState(const RateControlInput& input,
                                  int64_t now_ms) {
  switch (input.bw_state) {
    case BandwidthUsage::kBwNormal:
      if (rate_control_state_ == kRcHold) {
        time_last_bitrate_change_ = now_ms;
        rate_control_state_ = kRcIncrease;
      }
      break;
    case BandwidthUsage::kBwOverusing:
      if (rate_control_state_ != kRcDecrease) {
        rate_control_state_ = kRcDecrease;
      }
      break;
    case BandwidthUsage::kBwUnderusing:
      rate_control_state_ = kRcHold;
      break;
    default:
      assert(false);
  }
}

}  // namespace webrtc