/*
 *  Copyright (c) 2013 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.
 */

// Test to verify correct stereo and multi-channel operation.

#include <algorithm>
#include <string>
#include <list>

#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
#include "webrtc/test/testsupport/fileutils.h"

namespace webrtc {

struct TestParameters {
  int frame_size;
  int sample_rate;
  size_t num_channels;
};

// This is a parameterized test. The test parameters are supplied through a
// TestParameters struct, which is obtained through the GetParam() method.
//
// The objective of the test is to create a mono input signal and a
// multi-channel input signal, where each channel is identical to the mono
// input channel. The two input signals are processed through their respective
// NetEq instances. After that, the output signals are compared. The expected
// result is that each channel in the multi-channel output is identical to the
// mono output.
class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
 protected:
  static const int kTimeStepMs = 10;
  static const size_t kMaxBlockSize = 480;  // 10 ms @ 48 kHz.
  static const uint8_t kPayloadTypeMono = 95;
  static const uint8_t kPayloadTypeMulti = 96;

  NetEqStereoTest()
      : num_channels_(GetParam().num_channels),
        sample_rate_hz_(GetParam().sample_rate),
        samples_per_ms_(sample_rate_hz_ / 1000),
        frame_size_ms_(GetParam().frame_size),
        frame_size_samples_(
            static_cast<size_t>(frame_size_ms_ * samples_per_ms_)),
        output_size_samples_(10 * samples_per_ms_),
        rtp_generator_mono_(samples_per_ms_),
        rtp_generator_(samples_per_ms_),
        payload_size_bytes_(0),
        multi_payload_size_bytes_(0),
        last_send_time_(0),
        last_arrival_time_(0) {
    NetEq::Config config;
    config.sample_rate_hz = sample_rate_hz_;
    neteq_mono_ = NetEq::Create(config);
    neteq_ = NetEq::Create(config);
    input_ = new int16_t[frame_size_samples_];
    encoded_ = new uint8_t[2 * frame_size_samples_];
    input_multi_channel_ = new int16_t[frame_size_samples_ * num_channels_];
    encoded_multi_channel_ = new uint8_t[frame_size_samples_ * 2 *
                                         num_channels_];
    output_multi_channel_ = new int16_t[kMaxBlockSize * num_channels_];
  }

  ~NetEqStereoTest() {
    delete neteq_mono_;
    delete neteq_;
    delete [] input_;
    delete [] encoded_;
    delete [] input_multi_channel_;
    delete [] encoded_multi_channel_;
    delete [] output_multi_channel_;
  }

  virtual void SetUp() {
    const std::string file_name =
        webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
    input_file_.reset(new test::InputAudioFile(file_name));
    NetEqDecoder mono_decoder;
    NetEqDecoder multi_decoder;
    switch (sample_rate_hz_) {
      case 8000:
        mono_decoder = NetEqDecoder::kDecoderPCM16B;
        if (num_channels_ == 2) {
          multi_decoder = NetEqDecoder::kDecoderPCM16B_2ch;
        } else if (num_channels_ == 5) {
          multi_decoder = NetEqDecoder::kDecoderPCM16B_5ch;
        } else {
          FAIL() << "Only 2 and 5 channels supported for 8000 Hz.";
        }
        break;
      case 16000:
        mono_decoder = NetEqDecoder::kDecoderPCM16Bwb;
        if (num_channels_ == 2) {
          multi_decoder = NetEqDecoder::kDecoderPCM16Bwb_2ch;
        } else {
          FAIL() << "More than 2 channels is not supported for 16000 Hz.";
        }
        break;
      case 32000:
        mono_decoder = NetEqDecoder::kDecoderPCM16Bswb32kHz;
        if (num_channels_ == 2) {
          multi_decoder = NetEqDecoder::kDecoderPCM16Bswb32kHz_2ch;
        } else {
          FAIL() << "More than 2 channels is not supported for 32000 Hz.";
        }
        break;
      case 48000:
        mono_decoder = NetEqDecoder::kDecoderPCM16Bswb48kHz;
        if (num_channels_ == 2) {
          multi_decoder = NetEqDecoder::kDecoderPCM16Bswb48kHz_2ch;
        } else {
          FAIL() << "More than 2 channels is not supported for 48000 Hz.";
        }
        break;
      default:
        FAIL() << "We shouldn't get here.";
    }
    ASSERT_EQ(NetEq::kOK, neteq_mono_->RegisterPayloadType(mono_decoder, "mono",
                                                           kPayloadTypeMono));
    ASSERT_EQ(NetEq::kOK,
              neteq_->RegisterPayloadType(multi_decoder, "multi-channel",
                                          kPayloadTypeMulti));
  }

  virtual void TearDown() {}

  int GetNewPackets() {
    if (!input_file_->Read(frame_size_samples_, input_)) {
      return -1;
    }
    payload_size_bytes_ = WebRtcPcm16b_Encode(input_, frame_size_samples_,
                                             encoded_);
    if (frame_size_samples_ * 2 != payload_size_bytes_) {
      return -1;
    }
    int next_send_time = rtp_generator_mono_.GetRtpHeader(kPayloadTypeMono,
                                                          frame_size_samples_,
                                                          &rtp_header_mono_);
    test::InputAudioFile::DuplicateInterleaved(input_, frame_size_samples_,
                                               num_channels_,
                                               input_multi_channel_);
    multi_payload_size_bytes_ = WebRtcPcm16b_Encode(
        input_multi_channel_, frame_size_samples_ * num_channels_,
        encoded_multi_channel_);
    if (frame_size_samples_ * 2 * num_channels_ != multi_payload_size_bytes_) {
      return -1;
    }
    rtp_generator_.GetRtpHeader(kPayloadTypeMulti, frame_size_samples_,
                                &rtp_header_);
    return next_send_time;
  }

  void VerifyOutput(size_t num_samples) {
    for (size_t i = 0; i < num_samples; ++i) {
      for (size_t j = 0; j < num_channels_; ++j) {
        ASSERT_EQ(output_[i], output_multi_channel_[i * num_channels_ + j]) <<
            "Diff in sample " << i << ", channel " << j << ".";
      }
    }
  }

  virtual int GetArrivalTime(int send_time) {
    int arrival_time = last_arrival_time_ + (send_time - last_send_time_);
    last_send_time_ = send_time;
    last_arrival_time_ = arrival_time;
    return arrival_time;
  }

  virtual bool Lost() { return false; }

  void RunTest(int num_loops) {
    // Get next input packets (mono and multi-channel).
    int next_send_time;
    int next_arrival_time;
    do {
      next_send_time = GetNewPackets();
      ASSERT_NE(-1, next_send_time);
      next_arrival_time = GetArrivalTime(next_send_time);
    } while (Lost());  // If lost, immediately read the next packet.

    int time_now = 0;
    for (int k = 0; k < num_loops; ++k) {
      while (time_now >= next_arrival_time) {
        // Insert packet in mono instance.
        ASSERT_EQ(NetEq::kOK,
                  neteq_mono_->InsertPacket(rtp_header_mono_,
                                            rtc::ArrayView<const uint8_t>(
                                                encoded_, payload_size_bytes_),
                                            next_arrival_time));
        // Insert packet in multi-channel instance.
        ASSERT_EQ(NetEq::kOK, neteq_->InsertPacket(
                                  rtp_header_, rtc::ArrayView<const uint8_t>(
                                                   encoded_multi_channel_,
                                                   multi_payload_size_bytes_),
                                  next_arrival_time));
        // Get next input packets (mono and multi-channel).
        do {
          next_send_time = GetNewPackets();
          ASSERT_NE(-1, next_send_time);
          next_arrival_time = GetArrivalTime(next_send_time);
        } while (Lost());  // If lost, immediately read the next packet.
      }
      NetEqOutputType output_type;
      // Get audio from mono instance.
      size_t samples_per_channel;
      size_t num_channels;
      EXPECT_EQ(NetEq::kOK,
                neteq_mono_->GetAudio(kMaxBlockSize, output_,
                                      &samples_per_channel, &num_channels,
                                      &output_type));
      EXPECT_EQ(1u, num_channels);
      EXPECT_EQ(output_size_samples_, samples_per_channel);
      // Get audio from multi-channel instance.
      ASSERT_EQ(NetEq::kOK,
                neteq_->GetAudio(kMaxBlockSize * num_channels_,
                                 output_multi_channel_,
                                 &samples_per_channel, &num_channels,
                                 &output_type));
      EXPECT_EQ(num_channels_, num_channels);
      EXPECT_EQ(output_size_samples_, samples_per_channel);
      std::ostringstream ss;
      ss << "Lap number " << k << ".";
      SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
      // Compare mono and multi-channel.
      ASSERT_NO_FATAL_FAILURE(VerifyOutput(output_size_samples_));

      time_now += kTimeStepMs;
    }
  }

  const size_t num_channels_;
  const int sample_rate_hz_;
  const int samples_per_ms_;
  const int frame_size_ms_;
  const size_t frame_size_samples_;
  const size_t output_size_samples_;
  NetEq* neteq_mono_;
  NetEq* neteq_;
  test::RtpGenerator rtp_generator_mono_;
  test::RtpGenerator rtp_generator_;
  int16_t* input_;
  int16_t* input_multi_channel_;
  uint8_t* encoded_;
  uint8_t* encoded_multi_channel_;
  int16_t output_[kMaxBlockSize];
  int16_t* output_multi_channel_;
  WebRtcRTPHeader rtp_header_mono_;
  WebRtcRTPHeader rtp_header_;
  size_t payload_size_bytes_;
  size_t multi_payload_size_bytes_;
  int last_send_time_;
  int last_arrival_time_;
  rtc::scoped_ptr<test::InputAudioFile> input_file_;
};

class NetEqStereoTestNoJitter : public NetEqStereoTest {
 protected:
  NetEqStereoTestNoJitter()
      : NetEqStereoTest() {
    // Start the sender 100 ms before the receiver to pre-fill the buffer.
    // This is to avoid doing preemptive expand early in the test.
    // TODO(hlundin): Mock the decision making instead to control the modes.
    last_arrival_time_ = -100;
  }
};

#if defined(WEBRTC_ANDROID)
#define MAYBE_RunTest DISABLED_RunTest
#else
#define MAYBE_RunTest RunTest
#endif
TEST_P(NetEqStereoTestNoJitter, MAYBE_RunTest) {
  RunTest(8);
}

class NetEqStereoTestPositiveDrift : public NetEqStereoTest {
 protected:
  NetEqStereoTestPositiveDrift()
      : NetEqStereoTest(),
        drift_factor(0.9) {
    // Start the sender 100 ms before the receiver to pre-fill the buffer.
    // This is to avoid doing preemptive expand early in the test.
    // TODO(hlundin): Mock the decision making instead to control the modes.
    last_arrival_time_ = -100;
  }
  virtual int GetArrivalTime(int send_time) {
    int arrival_time = last_arrival_time_ +
        drift_factor * (send_time - last_send_time_);
    last_send_time_ = send_time;
    last_arrival_time_ = arrival_time;
    return arrival_time;
  }

  double drift_factor;
};

TEST_P(NetEqStereoTestPositiveDrift, MAYBE_RunTest) {
  RunTest(100);
}

class NetEqStereoTestNegativeDrift : public NetEqStereoTestPositiveDrift {
 protected:
  NetEqStereoTestNegativeDrift()
      : NetEqStereoTestPositiveDrift() {
    drift_factor = 1.1;
    last_arrival_time_ = 0;
  }
};

TEST_P(NetEqStereoTestNegativeDrift, MAYBE_RunTest) {
  RunTest(100);
}

class NetEqStereoTestDelays : public NetEqStereoTest {
 protected:
  static const int kDelayInterval = 10;
  static const int kDelay = 1000;
  NetEqStereoTestDelays()
      : NetEqStereoTest(),
        frame_index_(0) {
  }

  virtual int GetArrivalTime(int send_time) {
    // Deliver immediately, unless we have a back-log.
    int arrival_time = std::min(last_arrival_time_, send_time);
    if (++frame_index_ % kDelayInterval == 0) {
      // Delay this packet.
      arrival_time += kDelay;
    }
    last_send_time_ = send_time;
    last_arrival_time_ = arrival_time;
    return arrival_time;
  }

  int frame_index_;
};

TEST_P(NetEqStereoTestDelays, MAYBE_RunTest) {
  RunTest(1000);
}

class NetEqStereoTestLosses : public NetEqStereoTest {
 protected:
  static const int kLossInterval = 10;
  NetEqStereoTestLosses()
      : NetEqStereoTest(),
        frame_index_(0) {
  }

  virtual bool Lost() {
    return (++frame_index_) % kLossInterval == 0;
  }

  int frame_index_;
};

// TODO(pbos): Enable on non-Android, this went failing while being accidentally
// disabled on all platforms and not just Android.
// https://bugs.chromium.org/p/webrtc/issues/detail?id=5387
TEST_P(NetEqStereoTestLosses, DISABLED_RunTest) {
  RunTest(100);
}


// Creates a list of parameter sets.
std::list<TestParameters> GetTestParameters() {
  std::list<TestParameters> l;
  const int sample_rates[] = {8000, 16000, 32000};
  const int num_rates = sizeof(sample_rates) / sizeof(sample_rates[0]);
  // Loop through sample rates.
  for (int rate_index = 0; rate_index < num_rates; ++rate_index) {
    int sample_rate = sample_rates[rate_index];
    // Loop through all frame sizes between 10 and 60 ms.
    for (int frame_size = 10; frame_size <= 60; frame_size += 10) {
      TestParameters p;
      p.frame_size = frame_size;
      p.sample_rate = sample_rate;
      p.num_channels = 2;
      l.push_back(p);
      if (sample_rate == 8000) {
        // Add a five-channel test for 8000 Hz.
        p.num_channels = 5;
        l.push_back(p);
      }
    }
  }
  return l;
}

// Pretty-printing the test parameters in case of an error.
void PrintTo(const TestParameters& p, ::std::ostream* os) {
  *os << "{frame_size = " << p.frame_size <<
      ", num_channels = " << p.num_channels <<
      ", sample_rate = " << p.sample_rate << "}";
}

// Instantiate the tests. Each test is instantiated using the function above,
// so that all different parameter combinations are tested.
INSTANTIATE_TEST_CASE_P(MultiChannel,
                        NetEqStereoTestNoJitter,
                        ::testing::ValuesIn(GetTestParameters()));

INSTANTIATE_TEST_CASE_P(MultiChannel,
                        NetEqStereoTestPositiveDrift,
                        ::testing::ValuesIn(GetTestParameters()));

INSTANTIATE_TEST_CASE_P(MultiChannel,
                        NetEqStereoTestNegativeDrift,
                        ::testing::ValuesIn(GetTestParameters()));

INSTANTIATE_TEST_CASE_P(MultiChannel,
                        NetEqStereoTestDelays,
                        ::testing::ValuesIn(GetTestParameters()));

INSTANTIATE_TEST_CASE_P(MultiChannel,
                        NetEqStereoTestLosses,
                        ::testing::ValuesIn(GetTestParameters()));

}  // namespace webrtc