/*
 *  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 <math.h>
#include <stdio.h>

#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h"
#include "webrtc/modules/audio_coding/neteq/tools/output_audio_file.h"
#include "webrtc/modules/audio_coding/neteq/tools/output_wav_file.h"
#include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
#include "webrtc/test/testsupport/fileutils.h"

namespace webrtc {
namespace test {

const uint8_t kPayloadType = 95;
const int kOutputSizeMs = 10;
const int kInitSeed = 0x12345678;
const int kPacketLossTimeUnitMs = 10;

// Common validator for file names.
static bool ValidateFilename(const std::string& value, bool write) {
  FILE* fid = write ? fopen(value.c_str(), "wb") : fopen(value.c_str(), "rb");
  if (fid == nullptr)
    return false;
  fclose(fid);
  return true;
}

// Define switch for input file name.
static bool ValidateInFilename(const char* flagname, const std::string& value) {
  if (!ValidateFilename(value, false)) {
    printf("Invalid input filename.");
    return false;
  }
  return true;
}

DEFINE_string(
    in_filename,
    ResourcePath("audio_coding/speech_mono_16kHz", "pcm"),
    "Filename for input audio (specify sample rate with --input_sample_rate ,"
    "and channels with --channels).");

static const bool in_filename_dummy =
    RegisterFlagValidator(&FLAGS_in_filename, &ValidateInFilename);

// Define switch for sample rate.
static bool ValidateSampleRate(const char* flagname, int32_t value) {
  if (value == 8000 || value == 16000 || value == 32000 || value == 48000)
    return true;
  printf("Invalid sample rate should be 8000, 16000, 32000 or 48000 Hz.");
  return false;
}

DEFINE_int32(input_sample_rate, 16000, "Sample rate of input file in Hz.");

static const bool sample_rate_dummy =
    RegisterFlagValidator(&FLAGS_input_sample_rate, &ValidateSampleRate);

// Define switch for channels.
static bool ValidateChannels(const char* flagname, int32_t value) {
  if (value == 1)
    return true;
  printf("Invalid number of channels, current support only 1.");
  return false;
}

DEFINE_int32(channels, 1, "Number of channels in input audio.");

static const bool channels_dummy =
    RegisterFlagValidator(&FLAGS_channels, &ValidateChannels);

// Define switch for output file name.
static bool ValidateOutFilename(const char* flagname,
                                const std::string& value) {
  if (!ValidateFilename(value, true)) {
    printf("Invalid output filename.");
    return false;
  }
  return true;
}

DEFINE_string(out_filename,
              OutputPath() + "neteq_quality_test_out.pcm",
              "Name of output audio file.");

static const bool out_filename_dummy =
    RegisterFlagValidator(&FLAGS_out_filename, &ValidateOutFilename);

// Define switch for packet loss rate.
static bool ValidatePacketLossRate(const char* /* flag_name */, int32_t value) {
  if (value >= 0 && value <= 100)
    return true;
  printf("Invalid packet loss percentile, should be between 0 and 100.");
  return false;
}

// Define switch for runtime.
static bool ValidateRuntime(const char* flagname, int32_t value) {
  if (value > 0)
    return true;
  printf("Invalid runtime, should be greater than 0.");
  return false;
}

DEFINE_int32(runtime_ms, 10000, "Simulated runtime (milliseconds).");

static const bool runtime_dummy =
    RegisterFlagValidator(&FLAGS_runtime_ms, &ValidateRuntime);

DEFINE_int32(packet_loss_rate, 10, "Percentile of packet loss.");

static const bool packet_loss_rate_dummy =
    RegisterFlagValidator(&FLAGS_packet_loss_rate, &ValidatePacketLossRate);

// Define switch for random loss mode.
static bool ValidateRandomLossMode(const char* /* flag_name */, int32_t value) {
  if (value >= 0 && value <= 2)
    return true;
  printf("Invalid random packet loss mode, should be between 0 and 2.");
  return false;
}

DEFINE_int32(random_loss_mode, 1,
    "Random loss mode: 0--no loss, 1--uniform loss, 2--Gilbert Elliot loss.");
static const bool random_loss_mode_dummy =
    RegisterFlagValidator(&FLAGS_random_loss_mode, &ValidateRandomLossMode);

// Define switch for burst length.
static bool ValidateBurstLength(const char* /* flag_name */, int32_t value) {
  if (value >= kPacketLossTimeUnitMs)
    return true;
  printf("Invalid burst length, should be greater than %d ms.",
         kPacketLossTimeUnitMs);
  return false;
}

DEFINE_int32(burst_length, 30,
    "Burst length in milliseconds, only valid for Gilbert Elliot loss.");

static const bool burst_length_dummy =
    RegisterFlagValidator(&FLAGS_burst_length, &ValidateBurstLength);

// Define switch for drift factor.
static bool ValidateDriftFactor(const char* /* flag_name */, double value) {
  if (value > -0.1)
    return true;
  printf("Invalid drift factor, should be greater than -0.1.");
  return false;
}

DEFINE_double(drift_factor, 0.0, "Time drift factor.");

static const bool drift_factor_dummy =
    RegisterFlagValidator(&FLAGS_drift_factor, &ValidateDriftFactor);

// ProbTrans00Solver() is to calculate the transition probability from no-loss
// state to itself in a modified Gilbert Elliot packet loss model. The result is
// to achieve the target packet loss rate |loss_rate|, when a packet is not
// lost only if all |units| drawings within the duration of the packet result in
// no-loss.
static double ProbTrans00Solver(int units, double loss_rate,
                                double prob_trans_10) {
  if (units == 1)
    return prob_trans_10 / (1.0f - loss_rate) - prob_trans_10;
// 0 == prob_trans_00 ^ (units - 1) + (1 - loss_rate) / prob_trans_10 *
//     prob_trans_00 - (1 - loss_rate) * (1 + 1 / prob_trans_10).
// There is a unique solution between 0.0 and 1.0, due to the monotonicity and
// an opposite sign at 0.0 and 1.0.
// For simplicity, we reformulate the equation as
//     f(x) = x ^ (units - 1) + a x + b.
// Its derivative is
//     f'(x) = (units - 1) x ^ (units - 2) + a.
// The derivative is strictly greater than 0 when x is between 0 and 1.
// We use Newton's method to solve the equation, iteration is
//     x(k+1) = x(k) - f(x) / f'(x);
  const double kPrecision = 0.001f;
  const int kIterations = 100;
  const double a = (1.0f - loss_rate) / prob_trans_10;
  const double b = (loss_rate - 1.0f) * (1.0f + 1.0f / prob_trans_10);
  double x = 0.0f;  // Starting point;
  double f = b;
  double f_p;
  int iter = 0;
  while ((f >= kPrecision || f <= -kPrecision) && iter < kIterations) {
    f_p = (units - 1.0f) * pow(x, units - 2) + a;
    x -= f / f_p;
    if (x > 1.0f) {
      x = 1.0f;
    } else if (x < 0.0f) {
      x = 0.0f;
    }
    f = pow(x, units - 1) + a * x + b;
    iter ++;
  }
  return x;
}

NetEqQualityTest::NetEqQualityTest(int block_duration_ms,
                                   int in_sampling_khz,
                                   int out_sampling_khz,
                                   NetEqDecoder decoder_type)
    : decoder_type_(decoder_type),
      channels_(static_cast<size_t>(FLAGS_channels)),
      decoded_time_ms_(0),
      decodable_time_ms_(0),
      drift_factor_(FLAGS_drift_factor),
      packet_loss_rate_(FLAGS_packet_loss_rate),
      block_duration_ms_(block_duration_ms),
      in_sampling_khz_(in_sampling_khz),
      out_sampling_khz_(out_sampling_khz),
      in_size_samples_(
          static_cast<size_t>(in_sampling_khz_ * block_duration_ms_)),
      payload_size_bytes_(0),
      max_payload_bytes_(0),
      in_file_(new ResampleInputAudioFile(FLAGS_in_filename,
                                          FLAGS_input_sample_rate,
                                          in_sampling_khz * 1000)),
      rtp_generator_(
          new RtpGenerator(in_sampling_khz_, 0, 0, decodable_time_ms_)),
      total_payload_size_bytes_(0) {
  const std::string out_filename = FLAGS_out_filename;
  const std::string log_filename = out_filename + ".log";
  log_file_.open(log_filename.c_str(), std::ofstream::out);
  RTC_CHECK(log_file_.is_open());

  if (out_filename.size() >= 4 &&
      out_filename.substr(out_filename.size() - 4) == ".wav") {
    // Open a wav file.
    output_.reset(
        new webrtc::test::OutputWavFile(out_filename, 1000 * out_sampling_khz));
  } else {
    // Open a pcm file.
    output_.reset(new webrtc::test::OutputAudioFile(out_filename));
  }

  NetEq::Config config;
  config.sample_rate_hz = out_sampling_khz_ * 1000;
  neteq_.reset(
      NetEq::Create(config, webrtc::CreateBuiltinAudioDecoderFactory()));
  max_payload_bytes_ = in_size_samples_ * channels_ * sizeof(int16_t);
  in_data_.reset(new int16_t[in_size_samples_ * channels_]);
}

NetEqQualityTest::~NetEqQualityTest() {
  log_file_.close();
}

bool NoLoss::Lost() {
  return false;
}

UniformLoss::UniformLoss(double loss_rate)
    : loss_rate_(loss_rate) {
}

bool UniformLoss::Lost() {
  int drop_this = rand();
  return (drop_this < loss_rate_ * RAND_MAX);
}

GilbertElliotLoss::GilbertElliotLoss(double prob_trans_11, double prob_trans_01)
    : prob_trans_11_(prob_trans_11),
      prob_trans_01_(prob_trans_01),
      lost_last_(false),
      uniform_loss_model_(new UniformLoss(0)) {
}

GilbertElliotLoss::~GilbertElliotLoss() {}

bool GilbertElliotLoss::Lost() {
  // Simulate bursty channel (Gilbert model).
  // (1st order) Markov chain model with memory of the previous/last
  // packet state (lost or received).
  if (lost_last_) {
    // Previous packet was not received.
    uniform_loss_model_->set_loss_rate(prob_trans_11_);
    return lost_last_ = uniform_loss_model_->Lost();
  } else {
    uniform_loss_model_->set_loss_rate(prob_trans_01_);
    return lost_last_ = uniform_loss_model_->Lost();
  }
}

void NetEqQualityTest::SetUp() {
  ASSERT_EQ(0,
            neteq_->RegisterPayloadType(decoder_type_, "noname", kPayloadType));
  rtp_generator_->set_drift_factor(drift_factor_);

  int units = block_duration_ms_ / kPacketLossTimeUnitMs;
  switch (FLAGS_random_loss_mode) {
    case 1: {
      // |unit_loss_rate| is the packet loss rate for each unit time interval
      // (kPacketLossTimeUnitMs). Since a packet loss event is generated if any
      // of |block_duration_ms_ / kPacketLossTimeUnitMs| unit time intervals of
      // a full packet duration is drawn with a loss, |unit_loss_rate| fulfills
      // (1 - unit_loss_rate) ^ (block_duration_ms_ / kPacketLossTimeUnitMs) ==
      // 1 - packet_loss_rate.
      double unit_loss_rate = (1.0f - pow(1.0f - 0.01f * packet_loss_rate_,
          1.0f / units));
      loss_model_.reset(new UniformLoss(unit_loss_rate));
      break;
    }
    case 2: {
      // |FLAGS_burst_length| should be integer times of kPacketLossTimeUnitMs.
      ASSERT_EQ(0, FLAGS_burst_length % kPacketLossTimeUnitMs);

      // We do not allow 100 percent packet loss in Gilbert Elliot model, which
      // makes no sense.
      ASSERT_GT(100, packet_loss_rate_);

      // To guarantee the overall packet loss rate, transition probabilities
      // need to satisfy:
      // pi_0 * (1 - prob_trans_01_) ^ units +
      //     pi_1 * prob_trans_10_ ^ (units - 1) == 1 - loss_rate
      // pi_0 = prob_trans_10 / (prob_trans_10 + prob_trans_01_)
      //     is the stationary state probability of no-loss
      // pi_1 = prob_trans_01_ / (prob_trans_10 + prob_trans_01_)
      //     is the stationary state probability of loss
      // After a derivation prob_trans_00 should satisfy:
      // prob_trans_00 ^ (units - 1) = (loss_rate - 1) / prob_trans_10 *
      //     prob_trans_00 + (1 - loss_rate) * (1 + 1 / prob_trans_10).
      double loss_rate = 0.01f * packet_loss_rate_;
      double prob_trans_10 = 1.0f * kPacketLossTimeUnitMs / FLAGS_burst_length;
      double prob_trans_00 = ProbTrans00Solver(units, loss_rate, prob_trans_10);
      loss_model_.reset(new GilbertElliotLoss(1.0f - prob_trans_10,
                                              1.0f - prob_trans_00));
      break;
    }
    default: {
      loss_model_.reset(new NoLoss);
      break;
    }
  }

  // Make sure that the packet loss profile is same for all derived tests.
  srand(kInitSeed);
}

std::ofstream& NetEqQualityTest::Log() {
  return log_file_;
}

bool NetEqQualityTest::PacketLost() {
  int cycles = block_duration_ms_ / kPacketLossTimeUnitMs;

  // The loop is to make sure that codecs with different block lengths share the
  // same packet loss profile.
  bool lost = false;
  for (int idx = 0; idx < cycles; idx ++) {
    if (loss_model_->Lost()) {
      // The packet will be lost if any of the drawings indicates a loss, but
      // the loop has to go on to make sure that codecs with different block
      // lengths keep the same pace.
      lost = true;
    }
  }
  return lost;
}

int NetEqQualityTest::Transmit() {
  int packet_input_time_ms =
      rtp_generator_->GetRtpHeader(kPayloadType, in_size_samples_,
                                   &rtp_header_);
  Log() << "Packet of size "
        << payload_size_bytes_
        << " bytes, for frame at "
        << packet_input_time_ms
        << " ms ";
  if (payload_size_bytes_ > 0) {
    if (!PacketLost()) {
      int ret = neteq_->InsertPacket(
          rtp_header_,
          rtc::ArrayView<const uint8_t>(payload_.data(), payload_size_bytes_),
          packet_input_time_ms * in_sampling_khz_);
      if (ret != NetEq::kOK)
        return -1;
      Log() << "was sent.";
    } else {
      Log() << "was lost.";
    }
  }
  Log() << std::endl;
  return packet_input_time_ms;
}

int NetEqQualityTest::DecodeBlock() {
  bool muted;
  int ret = neteq_->GetAudio(&out_frame_, &muted);
  RTC_CHECK(!muted);

  if (ret != NetEq::kOK) {
    return -1;
  } else {
    RTC_DCHECK_EQ(out_frame_.num_channels_, channels_);
    RTC_DCHECK_EQ(out_frame_.samples_per_channel_,
                  static_cast<size_t>(kOutputSizeMs * out_sampling_khz_));
    RTC_CHECK(output_->WriteArray(
        out_frame_.data_,
        out_frame_.samples_per_channel_ * out_frame_.num_channels_));
    return static_cast<int>(out_frame_.samples_per_channel_);
  }
}

void NetEqQualityTest::Simulate() {
  int audio_size_samples;

  while (decoded_time_ms_ < FLAGS_runtime_ms) {
    // Assume 10 packets in packets buffer.
    while (decodable_time_ms_ - 10 * block_duration_ms_ < decoded_time_ms_) {
      ASSERT_TRUE(in_file_->Read(in_size_samples_ * channels_, &in_data_[0]));
      payload_.Clear();
      payload_size_bytes_ = EncodeBlock(&in_data_[0],
                                        in_size_samples_, &payload_,
                                        max_payload_bytes_);
      total_payload_size_bytes_ += payload_size_bytes_;
      decodable_time_ms_ = Transmit() + block_duration_ms_;
    }
    audio_size_samples = DecodeBlock();
    if (audio_size_samples > 0) {
      decoded_time_ms_ += audio_size_samples / out_sampling_khz_;
    }
  }
  Log() << "Average bit rate was "
        << 8.0f * total_payload_size_bytes_ / FLAGS_runtime_ms
        << " kbps"
        << std::endl;
}

}  // namespace test
}  // namespace webrtc