webrtc/modules/audio_coding/neteq4/neteq_unittest.cc

/*
 *  Copyright (c) 2011 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.
 */

/*
 * This file includes unit tests for NetEQ.
 */

#include "webrtc/modules/audio_coding/neteq4/interface/neteq.h"

#include <math.h>
#include <stdlib.h>
#include <string.h>  // memset

#include <set>
#include <string>
#include <vector>

#include "gflags/gflags.h"
#include "gtest/gtest.h"
#include "webrtc/modules/audio_coding/neteq4/test/NETEQTEST_RTPpacket.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
#include "webrtc/typedefs.h"

DEFINE_bool(gen_ref, false, "Generate reference files.");

namespace webrtc {

static bool IsAllZero(const int16_t* buf, int buf_length) {
  bool all_zero = true;
  for (int n = 0; n < buf_length && all_zero; ++n)
    all_zero = buf[n] == 0;
  return all_zero;
}

static bool IsAllNonZero(const int16_t* buf, int buf_length) {
  bool all_non_zero = true;
  for (int n = 0; n < buf_length && all_non_zero; ++n)
    all_non_zero = buf[n] != 0;
  return all_non_zero;
}

class RefFiles {
 public:
  RefFiles(const std::string& input_file, const std::string& output_file);
  ~RefFiles();
  template<class T> void ProcessReference(const T& test_results);
  template<typename T, size_t n> void ProcessReference(
      const T (&test_results)[n],
      size_t length);
  template<typename T, size_t n> void WriteToFile(
      const T (&test_results)[n],
      size_t length);
  template<typename T, size_t n> void ReadFromFileAndCompare(
      const T (&test_results)[n],
      size_t length);
  void WriteToFile(const NetEqNetworkStatistics& stats);
  void ReadFromFileAndCompare(const NetEqNetworkStatistics& stats);
  void WriteToFile(const RtcpStatistics& stats);
  void ReadFromFileAndCompare(const RtcpStatistics& stats);

  FILE* input_fp_;
  FILE* output_fp_;
};

RefFiles::RefFiles(const std::string &input_file,
                   const std::string &output_file)
    : input_fp_(NULL),
      output_fp_(NULL) {
  if (!input_file.empty()) {
    input_fp_ = fopen(input_file.c_str(), "rb");
    EXPECT_TRUE(input_fp_ != NULL);
  }
  if (!output_file.empty()) {
    output_fp_ = fopen(output_file.c_str(), "wb");
    EXPECT_TRUE(output_fp_ != NULL);
  }
}

RefFiles::~RefFiles() {
  if (input_fp_) {
    EXPECT_EQ(EOF, fgetc(input_fp_));  // Make sure that we reached the end.
    fclose(input_fp_);
  }
  if (output_fp_) fclose(output_fp_);
}

template<class T>
void RefFiles::ProcessReference(const T& test_results) {
  WriteToFile(test_results);
  ReadFromFileAndCompare(test_results);
}

template<typename T, size_t n>
void RefFiles::ProcessReference(const T (&test_results)[n], size_t length) {
  WriteToFile(test_results, length);
  ReadFromFileAndCompare(test_results, length);
}

template<typename T, size_t n>
void RefFiles::WriteToFile(const T (&test_results)[n], size_t length) {
  if (output_fp_) {
    ASSERT_EQ(length, fwrite(&test_results, sizeof(T), length, output_fp_));
  }
}

template<typename T, size_t n>
void RefFiles::ReadFromFileAndCompare(const T (&test_results)[n],
                                      size_t length) {
  if (input_fp_) {
    // Read from ref file.
    T* ref = new T[length];
    ASSERT_EQ(length, fread(ref, sizeof(T), length, input_fp_));
    // Compare
    ASSERT_EQ(0, memcmp(&test_results, ref, sizeof(T) * length));
    delete [] ref;
  }
}

void RefFiles::WriteToFile(const NetEqNetworkStatistics& stats) {
  if (output_fp_) {
    ASSERT_EQ(1u, fwrite(&stats, sizeof(NetEqNetworkStatistics), 1,
                         output_fp_));
  }
}

void RefFiles::ReadFromFileAndCompare(
    const NetEqNetworkStatistics& stats) {
  if (input_fp_) {
    // Read from ref file.
    size_t stat_size = sizeof(NetEqNetworkStatistics);
    NetEqNetworkStatistics ref_stats;
    ASSERT_EQ(1u, fread(&ref_stats, stat_size, 1, input_fp_));
    // Compare
    EXPECT_EQ(0, memcmp(&stats, &ref_stats, stat_size));
  }
}

void RefFiles::WriteToFile(const RtcpStatistics& stats) {
  if (output_fp_) {
    ASSERT_EQ(1u, fwrite(&(stats.fraction_lost), sizeof(stats.fraction_lost), 1,
                         output_fp_));
    ASSERT_EQ(1u, fwrite(&(stats.cumulative_lost),
                         sizeof(stats.cumulative_lost), 1, output_fp_));
    ASSERT_EQ(1u, fwrite(&(stats.extended_max_sequence_number),
                         sizeof(stats.extended_max_sequence_number), 1,
                         output_fp_));
    ASSERT_EQ(1u, fwrite(&(stats.jitter), sizeof(stats.jitter), 1,
                         output_fp_));
  }
}

void RefFiles::ReadFromFileAndCompare(
    const RtcpStatistics& stats) {
  if (input_fp_) {
    // Read from ref file.
    RtcpStatistics ref_stats;
    ASSERT_EQ(1u, fread(&(ref_stats.fraction_lost),
                        sizeof(ref_stats.fraction_lost), 1, input_fp_));
    ASSERT_EQ(1u, fread(&(ref_stats.cumulative_lost),
                        sizeof(ref_stats.cumulative_lost), 1, input_fp_));
    ASSERT_EQ(1u, fread(&(ref_stats.extended_max_sequence_number),
                        sizeof(ref_stats.extended_max_sequence_number), 1,
                        input_fp_));
    ASSERT_EQ(1u, fread(&(ref_stats.jitter), sizeof(ref_stats.jitter), 1,
                        input_fp_));
    // Compare
    EXPECT_EQ(ref_stats.fraction_lost, stats.fraction_lost);
    EXPECT_EQ(ref_stats.cumulative_lost, stats.cumulative_lost);
    EXPECT_EQ(ref_stats.extended_max_sequence_number,
              stats.extended_max_sequence_number);
    EXPECT_EQ(ref_stats.jitter, stats.jitter);
  }
}

class NetEqDecodingTest : public ::testing::Test {
 protected:
  // NetEQ must be polled for data once every 10 ms. Thus, neither of the
  // constants below can be changed.
  static const int kTimeStepMs = 10;
  static const int kBlockSize8kHz = kTimeStepMs * 8;
  static const int kBlockSize16kHz = kTimeStepMs * 16;
  static const int kBlockSize32kHz = kTimeStepMs * 32;
  static const int kMaxBlockSize = kBlockSize32kHz;
  static const int kInitSampleRateHz = 8000;

  NetEqDecodingTest();
  virtual void SetUp();
  virtual void TearDown();
  void SelectDecoders(NetEqDecoder* used_codec);
  void LoadDecoders();
  void OpenInputFile(const std::string &rtp_file);
  void Process(NETEQTEST_RTPpacket* rtp_ptr, int* out_len);
  void DecodeAndCompare(const std::string &rtp_file,
                        const std::string &ref_file);
  void DecodeAndCheckStats(const std::string &rtp_file,
                           const std::string &stat_ref_file,
                           const std::string &rtcp_ref_file);
  static void PopulateRtpInfo(int frame_index,
                              int timestamp,
                              WebRtcRTPHeader* rtp_info);
  static void PopulateCng(int frame_index,
                          int timestamp,
                          WebRtcRTPHeader* rtp_info,
                          uint8_t* payload,
                          int* payload_len);

  void CheckBgnOff(int sampling_rate, NetEqBackgroundNoiseMode bgn_mode);

  void WrapTest(uint16_t start_seq_no, uint32_t start_timestamp,
                const std::set<uint16_t>& drop_seq_numbers,
                bool expect_seq_no_wrap, bool expect_timestamp_wrap);

  void LongCngWithClockDrift(double drift_factor,
                             double network_freeze_ms,
                             bool pull_audio_during_freeze,
                             int delay_tolerance_ms,
                             int max_time_to_speech_ms);

  void DuplicateCng();

  NetEq* neteq_;
  FILE* rtp_fp_;
  unsigned int sim_clock_;
  int16_t out_data_[kMaxBlockSize];
  int output_sample_rate_;
};

// Allocating the static const so that it can be passed by reference.
const int NetEqDecodingTest::kTimeStepMs;
const int NetEqDecodingTest::kBlockSize8kHz;
const int NetEqDecodingTest::kBlockSize16kHz;
const int NetEqDecodingTest::kBlockSize32kHz;
const int NetEqDecodingTest::kMaxBlockSize;
const int NetEqDecodingTest::kInitSampleRateHz;

NetEqDecodingTest::NetEqDecodingTest()
    : neteq_(NULL),
      rtp_fp_(NULL),
      sim_clock_(0),
      output_sample_rate_(kInitSampleRateHz) {
  memset(out_data_, 0, sizeof(out_data_));
}

void NetEqDecodingTest::SetUp() {
  neteq_ = NetEq::Create(kInitSampleRateHz);
  ASSERT_TRUE(neteq_);
  LoadDecoders();
}

void NetEqDecodingTest::TearDown() {
  delete neteq_;
  if (rtp_fp_)
    fclose(rtp_fp_);
}

void NetEqDecodingTest::LoadDecoders() {
  // Load PCMu.
  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCMu, 0));
  // Load PCMa.
  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCMa, 8));
#ifndef WEBRTC_ANDROID
  // Load iLBC.
  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderILBC, 102));
#endif  // WEBRTC_ANDROID
  // Load iSAC.
  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderISAC, 103));
#ifndef WEBRTC_ANDROID
  // Load iSAC SWB.
  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderISACswb, 104));
  // Load iSAC FB.
  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderISACfb, 105));
#endif  // WEBRTC_ANDROID
  // Load PCM16B nb.
  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCM16B, 93));
  // Load PCM16B wb.
  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCM16Bwb, 94));
  // Load PCM16B swb32.
  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCM16Bswb32kHz, 95));
  // Load CNG 8 kHz.
  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderCNGnb, 13));
  // Load CNG 16 kHz.
  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderCNGwb, 98));
}

void NetEqDecodingTest::OpenInputFile(const std::string &rtp_file) {
  rtp_fp_ = fopen(rtp_file.c_str(), "rb");
  ASSERT_TRUE(rtp_fp_ != NULL);
  ASSERT_EQ(0, NETEQTEST_RTPpacket::skipFileHeader(rtp_fp_));
}

void NetEqDecodingTest::Process(NETEQTEST_RTPpacket* rtp, int* out_len) {
  // Check if time to receive.
  while ((sim_clock_ >= rtp->time()) &&
         (rtp->dataLen() >= 0)) {
    if (rtp->dataLen() > 0) {
      WebRtcRTPHeader rtpInfo;
      rtp->parseHeader(&rtpInfo);
      ASSERT_EQ(0, neteq_->InsertPacket(
          rtpInfo,
          rtp->payload(),
          rtp->payloadLen(),
          rtp->time() * (output_sample_rate_ / 1000)));
    }
    // Get next packet.
    ASSERT_NE(-1, rtp->readFromFile(rtp_fp_));
  }

  // Get audio from NetEq.
  NetEqOutputType type;
  int num_channels;
  ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, out_len,
                                &num_channels, &type));
  ASSERT_TRUE((*out_len == kBlockSize8kHz) ||
              (*out_len == kBlockSize16kHz) ||
              (*out_len == kBlockSize32kHz));
  output_sample_rate_ = *out_len / 10 * 1000;

  // Increase time.
  sim_clock_ += kTimeStepMs;
}

void NetEqDecodingTest::DecodeAndCompare(const std::string &rtp_file,
                                         const std::string &ref_file) {
  OpenInputFile(rtp_file);

  std::string ref_out_file = "";
  if (ref_file.empty()) {
    ref_out_file = webrtc::test::OutputPath() + "neteq_universal_ref.pcm";
  }
  RefFiles ref_files(ref_file, ref_out_file);

  NETEQTEST_RTPpacket rtp;
  ASSERT_GT(rtp.readFromFile(rtp_fp_), 0);
  int i = 0;
  while (rtp.dataLen() >= 0) {
    std::ostringstream ss;
    ss << "Lap number " << i++ << " in DecodeAndCompare while loop";
    SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
    int out_len = 0;
    ASSERT_NO_FATAL_FAILURE(Process(&rtp, &out_len));
    ASSERT_NO_FATAL_FAILURE(ref_files.ProcessReference(out_data_, out_len));
  }
}

void NetEqDecodingTest::DecodeAndCheckStats(const std::string &rtp_file,
                                            const std::string &stat_ref_file,
                                            const std::string &rtcp_ref_file) {
  OpenInputFile(rtp_file);
  std::string stat_out_file = "";
  if (stat_ref_file.empty()) {
    stat_out_file = webrtc::test::OutputPath() +
        "neteq_network_stats.dat";
  }
  RefFiles network_stat_files(stat_ref_file, stat_out_file);

  std::string rtcp_out_file = "";
  if (rtcp_ref_file.empty()) {
    rtcp_out_file = webrtc::test::OutputPath() +
        "neteq_rtcp_stats.dat";
  }
  RefFiles rtcp_stat_files(rtcp_ref_file, rtcp_out_file);

  NETEQTEST_RTPpacket rtp;
  ASSERT_GT(rtp.readFromFile(rtp_fp_), 0);
  while (rtp.dataLen() >= 0) {
    int out_len;
    Process(&rtp, &out_len);

    // Query the network statistics API once per second
    if (sim_clock_ % 1000 == 0) {
      // Process NetworkStatistics.
      NetEqNetworkStatistics network_stats;
      ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
      network_stat_files.ProcessReference(network_stats);

      // Process RTCPstat.
      RtcpStatistics rtcp_stats;
      neteq_->GetRtcpStatistics(&rtcp_stats);
      rtcp_stat_files.ProcessReference(rtcp_stats);
    }
  }
}

void NetEqDecodingTest::PopulateRtpInfo(int frame_index,
                                        int timestamp,
                                        WebRtcRTPHeader* rtp_info) {
  rtp_info->header.sequenceNumber = frame_index;
  rtp_info->header.timestamp = timestamp;
  rtp_info->header.ssrc = 0x1234;  // Just an arbitrary SSRC.
  rtp_info->header.payloadType = 94;  // PCM16b WB codec.
  rtp_info->header.markerBit = 0;
}

void NetEqDecodingTest::PopulateCng(int frame_index,
                                    int timestamp,
                                    WebRtcRTPHeader* rtp_info,
                                    uint8_t* payload,
                                    int* payload_len) {
  rtp_info->header.sequenceNumber = frame_index;
  rtp_info->header.timestamp = timestamp;
  rtp_info->header.ssrc = 0x1234;  // Just an arbitrary SSRC.
  rtp_info->header.payloadType = 98;  // WB CNG.
  rtp_info->header.markerBit = 0;
  payload[0] = 64;  // Noise level -64 dBov, quite arbitrarily chosen.
  *payload_len = 1;  // Only noise level, no spectral parameters.
}

void NetEqDecodingTest::CheckBgnOff(int sampling_rate_hz,
                                    NetEqBackgroundNoiseMode bgn_mode) {
  int expected_samples_per_channel = 0;
  uint8_t payload_type = 0xFF;  // Invalid.
  if (sampling_rate_hz == 8000) {
    expected_samples_per_channel = kBlockSize8kHz;
    payload_type = 93;  // PCM 16, 8 kHz.
  } else if (sampling_rate_hz == 16000) {
    expected_samples_per_channel = kBlockSize16kHz;
    payload_type = 94;  // PCM 16, 16 kHZ.
  } else if (sampling_rate_hz == 32000) {
    expected_samples_per_channel = kBlockSize32kHz;
    payload_type = 95;  // PCM 16, 32 kHz.
  } else {
    ASSERT_TRUE(false);  // Unsupported test case.
  }

  NetEqOutputType type;
  int16_t output[kBlockSize32kHz];  // Maximum size is chosen.
  int16_t input[kBlockSize32kHz];  // Maximum size is chosen.

  // Payload of 10 ms of PCM16 32 kHz.
  uint8_t payload[kBlockSize32kHz * sizeof(int16_t)];

  // Random payload.
  for (int n = 0; n < expected_samples_per_channel; ++n) {
    input[n] = (rand() & ((1 << 10) - 1)) - ((1 << 5) - 1);
  }
  int enc_len_bytes = WebRtcPcm16b_EncodeW16(
      input, expected_samples_per_channel, reinterpret_cast<int16_t*>(payload));
  ASSERT_EQ(enc_len_bytes, expected_samples_per_channel * 2);

  WebRtcRTPHeader rtp_info;
  PopulateRtpInfo(0, 0, &rtp_info);
  rtp_info.header.payloadType = payload_type;

  int number_channels = 0;
  int samples_per_channel = 0;

  uint32_t receive_timestamp = 0;
  for (int n = 0; n < 10; ++n) {  // Insert few packets and get audio.
    number_channels = 0;
    samples_per_channel = 0;
    ASSERT_EQ(0, neteq_->InsertPacket(
        rtp_info, payload, enc_len_bytes, receive_timestamp));
    ASSERT_EQ(0, neteq_->GetAudio(kBlockSize32kHz, output, &samples_per_channel,
                                  &number_channels, &type));
    ASSERT_EQ(1, number_channels);
    ASSERT_EQ(expected_samples_per_channel, samples_per_channel);
    ASSERT_EQ(kOutputNormal, type);

    // Next packet.
    rtp_info.header.timestamp += expected_samples_per_channel;
    rtp_info.header.sequenceNumber++;
    receive_timestamp += expected_samples_per_channel;
  }

  number_channels = 0;
  samples_per_channel = 0;

  // Get audio without inserting packets, expecting PLC and PLC-to-CNG. Pull one
  // frame without checking speech-type. This is the first frame pulled without
  // inserting any packet, and might not be labeled as PCL.
  ASSERT_EQ(0, neteq_->GetAudio(kBlockSize32kHz, output, &samples_per_channel,
                                &number_channels, &type));
  ASSERT_EQ(1, number_channels);
  ASSERT_EQ(expected_samples_per_channel, samples_per_channel);

  // To be able to test the fading of background noise we need at lease to pull
  // 610 frames.
  const int kFadingThreshold = 610;

  // Test several CNG-to-PLC packet for the expected behavior. The number 20 is
  // arbitrary, but sufficiently large to test enough number of frames.
  const int kNumPlcToCngTestFrames = 20;
  bool plc_to_cng = false;
  for (int n = 0; n < kFadingThreshold + kNumPlcToCngTestFrames; ++n) {
    number_channels = 0;
    samples_per_channel = 0;
    memset(output, 1, sizeof(output));  // Set to non-zero.
    ASSERT_EQ(0, neteq_->GetAudio(kBlockSize32kHz, output, &samples_per_channel,
                                  &number_channels, &type));
    ASSERT_EQ(1, number_channels);
    ASSERT_EQ(expected_samples_per_channel, samples_per_channel);
    if (type == kOutputPLCtoCNG) {
      plc_to_cng = true;
      double sum_squared = 0;
      for (int k = 0; k < number_channels * samples_per_channel; ++k)
        sum_squared += output[k] * output[k];
      if (bgn_mode == kBgnOn) {
        EXPECT_NE(0, sum_squared);
      } else if (bgn_mode == kBgnOff || n > kFadingThreshold) {
        EXPECT_EQ(0, sum_squared);
      }
    } else {
      EXPECT_EQ(kOutputPLC, type);
    }
  }
  EXPECT_TRUE(plc_to_cng);  // Just to be sure that PLC-to-CNG has occurred.
}

#if defined(_WIN32) && defined(WEBRTC_ARCH_64_BITS)
// Disabled for Windows 64-bit until webrtc:1458 is fixed.
#define MAYBE_TestBitExactness DISABLED_TestBitExactness
#else
#define MAYBE_TestBitExactness TestBitExactness
#endif

TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(MAYBE_TestBitExactness)) {
  const std::string input_rtp_file = webrtc::test::ProjectRootPath() +
      "resources/audio_coding/neteq_universal_new.rtp";
#if defined(_MSC_VER) && (_MSC_VER >= 1700)
  // For Visual Studio 2012 and later, we will have to use the generic reference
  // file, rather than the windows-specific one.
  const std::string input_ref_file = webrtc::test::ProjectRootPath() +
      "resources/audio_coding/neteq4_universal_ref.pcm";
#else
  const std::string input_ref_file =
      webrtc::test::ResourcePath("audio_coding/neteq4_universal_ref", "pcm");
#endif

  if (FLAGS_gen_ref) {
    DecodeAndCompare(input_rtp_file, "");
  } else {
    DecodeAndCompare(input_rtp_file, input_ref_file);
  }
}

TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(TestNetworkStatistics)) {
  const std::string input_rtp_file = webrtc::test::ProjectRootPath() +
      "resources/audio_coding/neteq_universal_new.rtp";
#if defined(_MSC_VER) && (_MSC_VER >= 1700)
  // For Visual Studio 2012 and later, we will have to use the generic reference
  // file, rather than the windows-specific one.
  const std::string network_stat_ref_file = webrtc::test::ProjectRootPath() +
      "resources/audio_coding/neteq4_network_stats.dat";
#else
  const std::string network_stat_ref_file =
      webrtc::test::ResourcePath("audio_coding/neteq4_network_stats", "dat");
#endif
  const std::string rtcp_stat_ref_file =
      webrtc::test::ResourcePath("audio_coding/neteq4_rtcp_stats", "dat");
  if (FLAGS_gen_ref) {
    DecodeAndCheckStats(input_rtp_file, "", "");
  } else {
    DecodeAndCheckStats(input_rtp_file, network_stat_ref_file,
                        rtcp_stat_ref_file);
  }
}

// TODO(hlundin): Re-enable test once the statistics interface is up and again.
TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(TestFrameWaitingTimeStatistics)) {
  // Use fax mode to avoid time-scaling. This is to simplify the testing of
  // packet waiting times in the packet buffer.
  neteq_->SetPlayoutMode(kPlayoutFax);
  ASSERT_EQ(kPlayoutFax, neteq_->PlayoutMode());
  // Insert 30 dummy packets at once. Each packet contains 10 ms 16 kHz audio.
  size_t num_frames = 30;
  const int kSamples = 10 * 16;
  const int kPayloadBytes = kSamples * 2;
  for (size_t i = 0; i < num_frames; ++i) {
    uint16_t payload[kSamples] = {0};
    WebRtcRTPHeader rtp_info;
    rtp_info.header.sequenceNumber = i;
    rtp_info.header.timestamp = i * kSamples;
    rtp_info.header.ssrc = 0x1234;  // Just an arbitrary SSRC.
    rtp_info.header.payloadType = 94;  // PCM16b WB codec.
    rtp_info.header.markerBit = 0;
    ASSERT_EQ(0, neteq_->InsertPacket(
        rtp_info,
        reinterpret_cast<uint8_t*>(payload),
        kPayloadBytes, 0));
  }
  // Pull out all data.
  for (size_t i = 0; i < num_frames; ++i) {
    int out_len;
    int num_channels;
    NetEqOutputType type;
    ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
                                  &num_channels, &type));
    ASSERT_EQ(kBlockSize16kHz, out_len);
  }

  std::vector<int> waiting_times;
  neteq_->WaitingTimes(&waiting_times);
  EXPECT_EQ(num_frames, waiting_times.size());
  // Since all frames are dumped into NetEQ at once, but pulled out with 10 ms
  // spacing (per definition), we expect the delay to increase with 10 ms for
  // each packet.
  for (size_t i = 0; i < waiting_times.size(); ++i) {
    EXPECT_EQ(static_cast<int>(i + 1) * 10, waiting_times[i]);
  }

  // Check statistics again and make sure it's been reset.
  neteq_->WaitingTimes(&waiting_times);
  int len = waiting_times.size();
  EXPECT_EQ(0, len);

  // Process > 100 frames, and make sure that that we get statistics
  // only for 100 frames. Note the new SSRC, causing NetEQ to reset.
  num_frames = 110;
  for (size_t i = 0; i < num_frames; ++i) {
    uint16_t payload[kSamples] = {0};
    WebRtcRTPHeader rtp_info;
    rtp_info.header.sequenceNumber = i;
    rtp_info.header.timestamp = i * kSamples;
    rtp_info.header.ssrc = 0x1235;  // Just an arbitrary SSRC.
    rtp_info.header.payloadType = 94;  // PCM16b WB codec.
    rtp_info.header.markerBit = 0;
    ASSERT_EQ(0, neteq_->InsertPacket(
        rtp_info,
        reinterpret_cast<uint8_t*>(payload),
        kPayloadBytes, 0));
    int out_len;
    int num_channels;
    NetEqOutputType type;
    ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
                                  &num_channels, &type));
    ASSERT_EQ(kBlockSize16kHz, out_len);
  }

  neteq_->WaitingTimes(&waiting_times);
  EXPECT_EQ(100u, waiting_times.size());
}

TEST_F(NetEqDecodingTest,
       DISABLED_ON_ANDROID(TestAverageInterArrivalTimeNegative)) {
  const int kNumFrames = 3000;  // Needed for convergence.
  int frame_index = 0;
  const int kSamples = 10 * 16;
  const int kPayloadBytes = kSamples * 2;
  while (frame_index < kNumFrames) {
    // Insert one packet each time, except every 10th time where we insert two
    // packets at once. This will create a negative clock-drift of approx. 10%.
    int num_packets = (frame_index % 10 == 0 ? 2 : 1);
    for (int n = 0; n < num_packets; ++n) {
      uint8_t payload[kPayloadBytes] = {0};
      WebRtcRTPHeader rtp_info;
      PopulateRtpInfo(frame_index, frame_index * kSamples, &rtp_info);
      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
      ++frame_index;
    }

    // Pull out data once.
    int out_len;
    int num_channels;
    NetEqOutputType type;
    ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
                                  &num_channels, &type));
    ASSERT_EQ(kBlockSize16kHz, out_len);
  }

  NetEqNetworkStatistics network_stats;
  ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
  EXPECT_EQ(-103196, network_stats.clockdrift_ppm);
}

TEST_F(NetEqDecodingTest,
       DISABLED_ON_ANDROID(TestAverageInterArrivalTimePositive)) {
  const int kNumFrames = 5000;  // Needed for convergence.
  int frame_index = 0;
  const int kSamples = 10 * 16;
  const int kPayloadBytes = kSamples * 2;
  for (int i = 0; i < kNumFrames; ++i) {
    // Insert one packet each time, except every 10th time where we don't insert
    // any packet. This will create a positive clock-drift of approx. 11%.
    int num_packets = (i % 10 == 9 ? 0 : 1);
    for (int n = 0; n < num_packets; ++n) {
      uint8_t payload[kPayloadBytes] = {0};
      WebRtcRTPHeader rtp_info;
      PopulateRtpInfo(frame_index, frame_index * kSamples, &rtp_info);
      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
      ++frame_index;
    }

    // Pull out data once.
    int out_len;
    int num_channels;
    NetEqOutputType type;
    ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
                                  &num_channels, &type));
    ASSERT_EQ(kBlockSize16kHz, out_len);
  }

  NetEqNetworkStatistics network_stats;
  ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
  EXPECT_EQ(110946, network_stats.clockdrift_ppm);
}

void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
                                              double network_freeze_ms,
                                              bool pull_audio_during_freeze,
                                              int delay_tolerance_ms,
                                              int max_time_to_speech_ms) {
  uint16_t seq_no = 0;
  uint32_t timestamp = 0;
  const int kFrameSizeMs = 30;
  const int kSamples = kFrameSizeMs * 16;
  const int kPayloadBytes = kSamples * 2;
  double next_input_time_ms = 0.0;
  double t_ms;
  int out_len;
  int num_channels;
  NetEqOutputType type;

  // Insert speech for 5 seconds.
  const int kSpeechDurationMs = 5000;
  for (t_ms = 0; t_ms < kSpeechDurationMs; t_ms += 10) {
    // Each turn in this for loop is 10 ms.
    while (next_input_time_ms <= t_ms) {
      // Insert one 30 ms speech frame.
      uint8_t payload[kPayloadBytes] = {0};
      WebRtcRTPHeader rtp_info;
      PopulateRtpInfo(seq_no, timestamp, &rtp_info);
      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
      ++seq_no;
      timestamp += kSamples;
      next_input_time_ms += static_cast<double>(kFrameSizeMs) * drift_factor;
    }
    // Pull out data once.
    ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
                                  &num_channels, &type));
    ASSERT_EQ(kBlockSize16kHz, out_len);
  }

  EXPECT_EQ(kOutputNormal, type);
  int32_t delay_before = timestamp - neteq_->PlayoutTimestamp();

  // Insert CNG for 1 minute (= 60000 ms).
  const int kCngPeriodMs = 100;
  const int kCngPeriodSamples = kCngPeriodMs * 16;  // Period in 16 kHz samples.
  const int kCngDurationMs = 60000;
  for (; t_ms < kSpeechDurationMs + kCngDurationMs; t_ms += 10) {
    // Each turn in this for loop is 10 ms.
    while (next_input_time_ms <= t_ms) {
      // Insert one CNG frame each 100 ms.
      uint8_t payload[kPayloadBytes];
      int payload_len;
      WebRtcRTPHeader rtp_info;
      PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, payload_len, 0));
      ++seq_no;
      timestamp += kCngPeriodSamples;
      next_input_time_ms += static_cast<double>(kCngPeriodMs) * drift_factor;
    }
    // Pull out data once.
    ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
                                  &num_channels, &type));
    ASSERT_EQ(kBlockSize16kHz, out_len);
  }

  EXPECT_EQ(kOutputCNG, type);

  if (network_freeze_ms > 0) {
    // First keep pulling audio for |network_freeze_ms| without inserting
    // any data, then insert CNG data corresponding to |network_freeze_ms|
    // without pulling any output audio.
    const double loop_end_time = t_ms + network_freeze_ms;
    for (; t_ms < loop_end_time; t_ms += 10) {
      // Pull out data once.
      ASSERT_EQ(0,
                neteq_->GetAudio(
                    kMaxBlockSize, out_data_, &out_len, &num_channels, &type));
      ASSERT_EQ(kBlockSize16kHz, out_len);
      EXPECT_EQ(kOutputCNG, type);
    }
    bool pull_once = pull_audio_during_freeze;
    // If |pull_once| is true, GetAudio will be called once half-way through
    // the network recovery period.
    double pull_time_ms = (t_ms + next_input_time_ms) / 2;
    while (next_input_time_ms <= t_ms) {
      if (pull_once && next_input_time_ms >= pull_time_ms) {
        pull_once = false;
        // Pull out data once.
        ASSERT_EQ(
            0,
            neteq_->GetAudio(
                kMaxBlockSize, out_data_, &out_len, &num_channels, &type));
        ASSERT_EQ(kBlockSize16kHz, out_len);
        EXPECT_EQ(kOutputCNG, type);
        t_ms += 10;
      }
      // Insert one CNG frame each 100 ms.
      uint8_t payload[kPayloadBytes];
      int payload_len;
      WebRtcRTPHeader rtp_info;
      PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, payload_len, 0));
      ++seq_no;
      timestamp += kCngPeriodSamples;
      next_input_time_ms += kCngPeriodMs * drift_factor;
    }
  }

  // Insert speech again until output type is speech.
  double speech_restart_time_ms = t_ms;
  while (type != kOutputNormal) {
    // Each turn in this for loop is 10 ms.
    while (next_input_time_ms <= t_ms) {
      // Insert one 30 ms speech frame.
      uint8_t payload[kPayloadBytes] = {0};
      WebRtcRTPHeader rtp_info;
      PopulateRtpInfo(seq_no, timestamp, &rtp_info);
      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
      ++seq_no;
      timestamp += kSamples;
      next_input_time_ms += kFrameSizeMs * drift_factor;
    }
    // Pull out data once.
    ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
                                  &num_channels, &type));
    ASSERT_EQ(kBlockSize16kHz, out_len);
    // Increase clock.
    t_ms += 10;
  }

  // Check that the speech starts again within reasonable time.
  double time_until_speech_returns_ms = t_ms - speech_restart_time_ms;
  EXPECT_LT(time_until_speech_returns_ms, max_time_to_speech_ms);
  int32_t delay_after = timestamp - neteq_->PlayoutTimestamp();
  // Compare delay before and after, and make sure it differs less than 20 ms.
  EXPECT_LE(delay_after, delay_before + delay_tolerance_ms * 16);
  EXPECT_GE(delay_after, delay_before - delay_tolerance_ms * 16);
}

TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(LongCngWithNegativeClockDrift)) {
  // Apply a clock drift of -25 ms / s (sender faster than receiver).
  const double kDriftFactor = 1000.0 / (1000.0 + 25.0);
  const double kNetworkFreezeTimeMs = 0.0;
  const bool kGetAudioDuringFreezeRecovery = false;
  const int kDelayToleranceMs = 20;
  const int kMaxTimeToSpeechMs = 100;
  LongCngWithClockDrift(kDriftFactor,
                        kNetworkFreezeTimeMs,
                        kGetAudioDuringFreezeRecovery,
                        kDelayToleranceMs,
                        kMaxTimeToSpeechMs);
}

TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(LongCngWithPositiveClockDrift)) {
  // Apply a clock drift of +25 ms / s (sender slower than receiver).
  const double kDriftFactor = 1000.0 / (1000.0 - 25.0);
  const double kNetworkFreezeTimeMs = 0.0;
  const bool kGetAudioDuringFreezeRecovery = false;
  const int kDelayToleranceMs = 20;
  const int kMaxTimeToSpeechMs = 100;
  LongCngWithClockDrift(kDriftFactor,
                        kNetworkFreezeTimeMs,
                        kGetAudioDuringFreezeRecovery,
                        kDelayToleranceMs,
                        kMaxTimeToSpeechMs);
}

TEST_F(NetEqDecodingTest,
       DISABLED_ON_ANDROID(LongCngWithNegativeClockDriftNetworkFreeze)) {
  // Apply a clock drift of -25 ms / s (sender faster than receiver).
  const double kDriftFactor = 1000.0 / (1000.0 + 25.0);
  const double kNetworkFreezeTimeMs = 5000.0;
  const bool kGetAudioDuringFreezeRecovery = false;
  const int kDelayToleranceMs = 50;
  const int kMaxTimeToSpeechMs = 200;
  LongCngWithClockDrift(kDriftFactor,
                        kNetworkFreezeTimeMs,
                        kGetAudioDuringFreezeRecovery,
                        kDelayToleranceMs,
                        kMaxTimeToSpeechMs);
}

TEST_F(NetEqDecodingTest,
       DISABLED_ON_ANDROID(LongCngWithPositiveClockDriftNetworkFreeze)) {
  // Apply a clock drift of +25 ms / s (sender slower than receiver).
  const double kDriftFactor = 1000.0 / (1000.0 - 25.0);
  const double kNetworkFreezeTimeMs = 5000.0;
  const bool kGetAudioDuringFreezeRecovery = false;
  const int kDelayToleranceMs = 20;
  const int kMaxTimeToSpeechMs = 100;
  LongCngWithClockDrift(kDriftFactor,
                        kNetworkFreezeTimeMs,
                        kGetAudioDuringFreezeRecovery,
                        kDelayToleranceMs,
                        kMaxTimeToSpeechMs);
}

TEST_F(
    NetEqDecodingTest,
    DISABLED_ON_ANDROID(LongCngWithPositiveClockDriftNetworkFreezeExtraPull)) {
  // Apply a clock drift of +25 ms / s (sender slower than receiver).
  const double kDriftFactor = 1000.0 / (1000.0 - 25.0);
  const double kNetworkFreezeTimeMs = 5000.0;
  const bool kGetAudioDuringFreezeRecovery = true;
  const int kDelayToleranceMs = 20;
  const int kMaxTimeToSpeechMs = 100;
  LongCngWithClockDrift(kDriftFactor,
                        kNetworkFreezeTimeMs,
                        kGetAudioDuringFreezeRecovery,
                        kDelayToleranceMs,
                        kMaxTimeToSpeechMs);
}

TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(LongCngWithoutClockDrift)) {
  const double kDriftFactor = 1.0;  // No drift.
  const double kNetworkFreezeTimeMs = 0.0;
  const bool kGetAudioDuringFreezeRecovery = false;
  const int kDelayToleranceMs = 10;
  const int kMaxTimeToSpeechMs = 50;
  LongCngWithClockDrift(kDriftFactor,
                        kNetworkFreezeTimeMs,
                        kGetAudioDuringFreezeRecovery,
                        kDelayToleranceMs,
                        kMaxTimeToSpeechMs);
}

TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(UnknownPayloadType)) {
  const int kPayloadBytes = 100;
  uint8_t payload[kPayloadBytes] = {0};
  WebRtcRTPHeader rtp_info;
  PopulateRtpInfo(0, 0, &rtp_info);
  rtp_info.header.payloadType = 1;  // Not registered as a decoder.
  EXPECT_EQ(NetEq::kFail,
            neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
  EXPECT_EQ(NetEq::kUnknownRtpPayloadType, neteq_->LastError());
}

TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(OversizePacket)) {
  // Payload size is greater than packet buffer size
  const int kPayloadBytes = NetEq::kMaxBytesInBuffer + 1;
  uint8_t payload[kPayloadBytes] = {0};
  WebRtcRTPHeader rtp_info;
  PopulateRtpInfo(0, 0, &rtp_info);
  rtp_info.header.payloadType = 103;  // iSAC, no packet splitting.
  EXPECT_EQ(NetEq::kFail,
            neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
  EXPECT_EQ(NetEq::kOversizePacket, neteq_->LastError());
}

TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(DecoderError)) {
  const int kPayloadBytes = 100;
  uint8_t payload[kPayloadBytes] = {0};
  WebRtcRTPHeader rtp_info;
  PopulateRtpInfo(0, 0, &rtp_info);
  rtp_info.header.payloadType = 103;  // iSAC, but the payload is invalid.
  EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
  NetEqOutputType type;
  // Set all of |out_data_| to 1, and verify that it was set to 0 by the call
  // to GetAudio.
  for (int i = 0; i < kMaxBlockSize; ++i) {
    out_data_[i] = 1;
  }
  int num_channels;
  int samples_per_channel;
  EXPECT_EQ(NetEq::kFail,
            neteq_->GetAudio(kMaxBlockSize, out_data_,
                             &samples_per_channel, &num_channels, &type));
  // Verify that there is a decoder error to check.
  EXPECT_EQ(NetEq::kDecoderErrorCode, neteq_->LastError());
  // Code 6730 is an iSAC error code.
  EXPECT_EQ(6730, neteq_->LastDecoderError());
  // Verify that the first 160 samples are set to 0, and that the remaining
  // samples are left unmodified.
  static const int kExpectedOutputLength = 160;  // 10 ms at 16 kHz sample rate.
  for (int i = 0; i < kExpectedOutputLength; ++i) {
    std::ostringstream ss;
    ss << "i = " << i;
    SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
    EXPECT_EQ(0, out_data_[i]);
  }
  for (int i = kExpectedOutputLength; i < kMaxBlockSize; ++i) {
    std::ostringstream ss;
    ss << "i = " << i;
    SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
    EXPECT_EQ(1, out_data_[i]);
  }
}

TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(GetAudioBeforeInsertPacket)) {
  NetEqOutputType type;
  // Set all of |out_data_| to 1, and verify that it was set to 0 by the call
  // to GetAudio.
  for (int i = 0; i < kMaxBlockSize; ++i) {
    out_data_[i] = 1;
  }
  int num_channels;
  int samples_per_channel;
  EXPECT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_,
                                &samples_per_channel,
                                &num_channels, &type));
  // Verify that the first block of samples is set to 0.
  static const int kExpectedOutputLength =
      kInitSampleRateHz / 100;  // 10 ms at initial sample rate.
  for (int i = 0; i < kExpectedOutputLength; ++i) {
    std::ostringstream ss;
    ss << "i = " << i;
    SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
    EXPECT_EQ(0, out_data_[i]);
  }
}

TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(BackgroundNoise)) {
  neteq_->SetBackgroundNoiseMode(kBgnOn);
  CheckBgnOff(8000, kBgnOn);
  CheckBgnOff(16000, kBgnOn);
  CheckBgnOff(32000, kBgnOn);
  EXPECT_EQ(kBgnOn, neteq_->BackgroundNoiseMode());

  neteq_->SetBackgroundNoiseMode(kBgnOff);
  CheckBgnOff(8000, kBgnOff);
  CheckBgnOff(16000, kBgnOff);
  CheckBgnOff(32000, kBgnOff);
  EXPECT_EQ(kBgnOff, neteq_->BackgroundNoiseMode());

  neteq_->SetBackgroundNoiseMode(kBgnFade);
  CheckBgnOff(8000, kBgnFade);
  CheckBgnOff(16000, kBgnFade);
  CheckBgnOff(32000, kBgnFade);
  EXPECT_EQ(kBgnFade, neteq_->BackgroundNoiseMode());
}

TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(SyncPacketInsert)) {
  WebRtcRTPHeader rtp_info;
  uint32_t receive_timestamp = 0;
  // For the readability use the following payloads instead of the defaults of
  // this test.
  uint8_t kPcm16WbPayloadType = 1;
  uint8_t kCngNbPayloadType = 2;
  uint8_t kCngWbPayloadType = 3;
  uint8_t kCngSwb32PayloadType = 4;
  uint8_t kCngSwb48PayloadType = 5;
  uint8_t kAvtPayloadType = 6;
  uint8_t kRedPayloadType = 7;
  uint8_t kIsacPayloadType = 9;  // Payload type 8 is already registered.

  // Register decoders.
  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCM16Bwb,
                                           kPcm16WbPayloadType));
  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderCNGnb, kCngNbPayloadType));
  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderCNGwb, kCngWbPayloadType));
  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderCNGswb32kHz,
                                           kCngSwb32PayloadType));
  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderCNGswb48kHz,
                                           kCngSwb48PayloadType));
  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderAVT, kAvtPayloadType));
  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderRED, kRedPayloadType));
  ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderISAC, kIsacPayloadType));

  PopulateRtpInfo(0, 0, &rtp_info);
  rtp_info.header.payloadType = kPcm16WbPayloadType;

  // The first packet injected cannot be sync-packet.
  EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));

  // Payload length of 10 ms PCM16 16 kHz.
  const int kPayloadBytes = kBlockSize16kHz * sizeof(int16_t);
  uint8_t payload[kPayloadBytes] = {0};
  ASSERT_EQ(0, neteq_->InsertPacket(
      rtp_info, payload, kPayloadBytes, receive_timestamp));

  // Next packet. Last packet contained 10 ms audio.
  rtp_info.header.sequenceNumber++;
  rtp_info.header.timestamp += kBlockSize16kHz;
  receive_timestamp += kBlockSize16kHz;

  // Unacceptable payload types CNG, AVT (DTMF), RED.
  rtp_info.header.payloadType = kCngNbPayloadType;
  EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));

  rtp_info.header.payloadType = kCngWbPayloadType;
  EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));

  rtp_info.header.payloadType = kCngSwb32PayloadType;
  EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));

  rtp_info.header.payloadType = kCngSwb48PayloadType;
  EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));

  rtp_info.header.payloadType = kAvtPayloadType;
  EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));

  rtp_info.header.payloadType = kRedPayloadType;
  EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));

  // Change of codec cannot be initiated with a sync packet.
  rtp_info.header.payloadType = kIsacPayloadType;
  EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));

  // Change of SSRC is not allowed with a sync packet.
  rtp_info.header.payloadType = kPcm16WbPayloadType;
  ++rtp_info.header.ssrc;
  EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));

  --rtp_info.header.ssrc;
  EXPECT_EQ(0, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
}

// First insert several noise like packets, then sync-packets. Decoding all
// packets should not produce error, statistics should not show any packet loss
// and sync-packets should decode to zero.
TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(SyncPacketDecode)) {
  WebRtcRTPHeader rtp_info;
  PopulateRtpInfo(0, 0, &rtp_info);
  const int kPayloadBytes = kBlockSize16kHz * sizeof(int16_t);
  uint8_t payload[kPayloadBytes];
  int16_t decoded[kBlockSize16kHz];
  for (int n = 0; n < kPayloadBytes; ++n) {
    payload[n] = (rand() & 0xF0) + 1;  // Non-zero random sequence.
  }
  // Insert some packets which decode to noise. We are not interested in
  // actual decoded values.
  NetEqOutputType output_type;
  int num_channels;
  int samples_per_channel;
  uint32_t receive_timestamp = 0;
  int delay_samples = 0;
  for (int n = 0; n < 100; ++n) {
    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
                                      receive_timestamp));
    ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
                                  &samples_per_channel, &num_channels,
                                  &output_type));
    ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
    ASSERT_EQ(1, num_channels);

    // Even if there is RTP packet in NetEq's buffer, the first frame pulled
    // from NetEq starts with few zero samples. Here we measure this delay.
    if (n == 0) {
      while (decoded[delay_samples] == 0) delay_samples++;
    }
    rtp_info.header.sequenceNumber++;
    rtp_info.header.timestamp += kBlockSize16kHz;
    receive_timestamp += kBlockSize16kHz;
  }
  const int kNumSyncPackets = 10;
  // Insert sync-packets, the decoded sequence should be all-zero.
  for (int n = 0; n < kNumSyncPackets; ++n) {
    ASSERT_EQ(0, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
    ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
                                  &samples_per_channel, &num_channels,
                                  &output_type));
    ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
    ASSERT_EQ(1, num_channels);
    EXPECT_TRUE(IsAllZero(&decoded[delay_samples],
                          samples_per_channel * num_channels - delay_samples));
    delay_samples = 0;  // Delay only matters in the first frame.
    rtp_info.header.sequenceNumber++;
    rtp_info.header.timestamp += kBlockSize16kHz;
    receive_timestamp += kBlockSize16kHz;
  }
  // We insert a regular packet, if sync packet are not correctly buffered then
  // network statistics would show some packet loss.
  ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
                                    receive_timestamp));
  ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
                                &samples_per_channel, &num_channels,
                                &output_type));
  // Make sure the last inserted packet is decoded and there are non-zero
  // samples.
  EXPECT_FALSE(IsAllZero(decoded, samples_per_channel * num_channels));
  NetEqNetworkStatistics network_stats;
  ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
  // Expecting a "clean" network.
  EXPECT_EQ(0, network_stats.packet_loss_rate);
  EXPECT_EQ(0, network_stats.expand_rate);
  EXPECT_EQ(0, network_stats.accelerate_rate);
  EXPECT_EQ(0, network_stats.preemptive_rate);
}

// Test if the size of the packet buffer reported correctly when containing
// sync packets. Also, test if network packets override sync packets. That is to
// prefer decoding a network packet to a sync packet, if both have same sequence
// number and timestamp.
TEST_F(NetEqDecodingTest,
       DISABLED_ON_ANDROID(SyncPacketBufferSizeAndOverridenByNetworkPackets)) {
  WebRtcRTPHeader rtp_info;
  PopulateRtpInfo(0, 0, &rtp_info);
  const int kPayloadBytes = kBlockSize16kHz * sizeof(int16_t);
  uint8_t payload[kPayloadBytes];
  int16_t decoded[kBlockSize16kHz];
  for (int n = 0; n < kPayloadBytes; ++n) {
    payload[n] = (rand() & 0xF0) + 1;  // Non-zero random sequence.
  }
  // Insert some packets which decode to noise. We are not interested in
  // actual decoded values.
  NetEqOutputType output_type;
  int num_channels;
  int samples_per_channel;
  uint32_t receive_timestamp = 0;
  for (int n = 0; n < 1; ++n) {
    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
                                      receive_timestamp));
    ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
                                  &samples_per_channel, &num_channels,
                                  &output_type));
    ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
    ASSERT_EQ(1, num_channels);
    rtp_info.header.sequenceNumber++;
    rtp_info.header.timestamp += kBlockSize16kHz;
    receive_timestamp += kBlockSize16kHz;
  }
  const int kNumSyncPackets = 10;

  WebRtcRTPHeader first_sync_packet_rtp_info;
  memcpy(&first_sync_packet_rtp_info, &rtp_info, sizeof(rtp_info));

  // Insert sync-packets, but no decoding.
  for (int n = 0; n < kNumSyncPackets; ++n) {
    ASSERT_EQ(0, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
    rtp_info.header.sequenceNumber++;
    rtp_info.header.timestamp += kBlockSize16kHz;
    receive_timestamp += kBlockSize16kHz;
  }
  NetEqNetworkStatistics network_stats;
  ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
  EXPECT_EQ(kNumSyncPackets * 10, network_stats.current_buffer_size_ms);

  // Rewind |rtp_info| to that of the first sync packet.
  memcpy(&rtp_info, &first_sync_packet_rtp_info, sizeof(rtp_info));

  // Insert.
  for (int n = 0; n < kNumSyncPackets; ++n) {
    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
                                      receive_timestamp));
    rtp_info.header.sequenceNumber++;
    rtp_info.header.timestamp += kBlockSize16kHz;
    receive_timestamp += kBlockSize16kHz;
  }

  // Decode.
  for (int n = 0; n < kNumSyncPackets; ++n) {
    ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
                                  &samples_per_channel, &num_channels,
                                  &output_type));
    ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
    ASSERT_EQ(1, num_channels);
    EXPECT_TRUE(IsAllNonZero(decoded, samples_per_channel * num_channels));
  }
}

void NetEqDecodingTest::WrapTest(uint16_t start_seq_no,
                                 uint32_t start_timestamp,
                                 const std::set<uint16_t>& drop_seq_numbers,
                                 bool expect_seq_no_wrap,
                                 bool expect_timestamp_wrap) {
  uint16_t seq_no = start_seq_no;
  uint32_t timestamp = start_timestamp;
  const int kBlocksPerFrame = 3;  // Number of 10 ms blocks per frame.
  const int kFrameSizeMs = kBlocksPerFrame * kTimeStepMs;
  const int kSamples = kBlockSize16kHz * kBlocksPerFrame;
  const int kPayloadBytes = kSamples * sizeof(int16_t);
  double next_input_time_ms = 0.0;
  int16_t decoded[kBlockSize16kHz];
  int num_channels;
  int samples_per_channel;
  NetEqOutputType output_type;
  uint32_t receive_timestamp = 0;

  // Insert speech for 2 seconds.
  const int kSpeechDurationMs = 2000;
  int packets_inserted = 0;
  uint16_t last_seq_no;
  uint32_t last_timestamp;
  bool timestamp_wrapped = false;
  bool seq_no_wrapped = false;
  for (double t_ms = 0; t_ms < kSpeechDurationMs; t_ms += 10) {
    // Each turn in this for loop is 10 ms.
    while (next_input_time_ms <= t_ms) {
      // Insert one 30 ms speech frame.
      uint8_t payload[kPayloadBytes] = {0};
      WebRtcRTPHeader rtp_info;
      PopulateRtpInfo(seq_no, timestamp, &rtp_info);
      if (drop_seq_numbers.find(seq_no) == drop_seq_numbers.end()) {
        // This sequence number was not in the set to drop. Insert it.
        ASSERT_EQ(0,
                  neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
                                       receive_timestamp));
        ++packets_inserted;
      }
      NetEqNetworkStatistics network_stats;
      ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));

      // Due to internal NetEq logic, preferred buffer-size is about 4 times the
      // packet size for first few packets. Therefore we refrain from checking
      // the criteria.
      if (packets_inserted > 4) {
        // Expect preferred and actual buffer size to be no more than 2 frames.
        EXPECT_LE(network_stats.preferred_buffer_size_ms, kFrameSizeMs * 2);
        EXPECT_LE(network_stats.current_buffer_size_ms, kFrameSizeMs * 2);
      }
      last_seq_no = seq_no;
      last_timestamp = timestamp;

      ++seq_no;
      timestamp += kSamples;
      receive_timestamp += kSamples;
      next_input_time_ms += static_cast<double>(kFrameSizeMs);

      seq_no_wrapped |= seq_no < last_seq_no;
      timestamp_wrapped |= timestamp < last_timestamp;
    }
    // Pull out data once.
    ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
                                  &samples_per_channel, &num_channels,
                                  &output_type));
    ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
    ASSERT_EQ(1, num_channels);

    // Expect delay (in samples) to be less than 2 packets.
    EXPECT_LE(timestamp - neteq_->PlayoutTimestamp(),
              static_cast<uint32_t>(kSamples * 2));
  }
  // Make sure we have actually tested wrap-around.
  ASSERT_EQ(expect_seq_no_wrap, seq_no_wrapped);
  ASSERT_EQ(expect_timestamp_wrap, timestamp_wrapped);
}

TEST_F(NetEqDecodingTest, SequenceNumberWrap) {
  // Start with a sequence number that will soon wrap.
  std::set<uint16_t> drop_seq_numbers;  // Don't drop any packets.
  WrapTest(0xFFFF - 10, 0, drop_seq_numbers, true, false);
}

TEST_F(NetEqDecodingTest, SequenceNumberWrapAndDrop) {
  // Start with a sequence number that will soon wrap.
  std::set<uint16_t> drop_seq_numbers;
  drop_seq_numbers.insert(0xFFFF);
  drop_seq_numbers.insert(0x0);
  WrapTest(0xFFFF - 10, 0, drop_seq_numbers, true, false);
}

TEST_F(NetEqDecodingTest, TimestampWrap) {
  // Start with a timestamp that will soon wrap.
  std::set<uint16_t> drop_seq_numbers;
  WrapTest(0, 0xFFFFFFFF - 3000, drop_seq_numbers, false, true);
}

TEST_F(NetEqDecodingTest, TimestampAndSequenceNumberWrap) {
  // Start with a timestamp and a sequence number that will wrap at the same
  // time.
  std::set<uint16_t> drop_seq_numbers;
  WrapTest(0xFFFF - 10, 0xFFFFFFFF - 5000, drop_seq_numbers, true, true);
}

void NetEqDecodingTest::DuplicateCng() {
  uint16_t seq_no = 0;
  uint32_t timestamp = 0;
  const int kFrameSizeMs = 10;
  const int kSampleRateKhz = 16;
  const int kSamples = kFrameSizeMs * kSampleRateKhz;
  const int kPayloadBytes = kSamples * 2;

  // Insert three speech packet. Three are needed to get the frame length
  // correct.
  int out_len;
  int num_channels;
  NetEqOutputType type;
  uint8_t payload[kPayloadBytes] = {0};
  WebRtcRTPHeader rtp_info;
  for (int i = 0; i < 3; ++i) {
    PopulateRtpInfo(seq_no, timestamp, &rtp_info);
    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
    ++seq_no;
    timestamp += kSamples;

    // Pull audio once.
    ASSERT_EQ(0,
              neteq_->GetAudio(
                  kMaxBlockSize, out_data_, &out_len, &num_channels, &type));
    ASSERT_EQ(kBlockSize16kHz, out_len);
  }
  // Verify speech output.
  EXPECT_EQ(kOutputNormal, type);

  // Insert same CNG packet twice.
  const int kCngPeriodMs = 100;
  const int kCngPeriodSamples = kCngPeriodMs * kSampleRateKhz;
  int payload_len;
  PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
  // This is the first time this CNG packet is inserted.
  ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, payload_len, 0));

  // Pull audio once and make sure CNG is played.
  ASSERT_EQ(0,
            neteq_->GetAudio(
                kMaxBlockSize, out_data_, &out_len, &num_channels, &type));
  ASSERT_EQ(kBlockSize16kHz, out_len);
  EXPECT_EQ(kOutputCNG, type);
  EXPECT_EQ(timestamp - 10, neteq_->PlayoutTimestamp());

  // Insert the same CNG packet again. Note that at this point it is old, since
  // we have already decoded the first copy of it.
  ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, payload_len, 0));

  // Pull audio until we have played |kCngPeriodMs| of CNG. Start at 10 ms since
  // we have already pulled out CNG once.
  for (int cng_time_ms = 10; cng_time_ms < kCngPeriodMs; cng_time_ms += 10) {
    ASSERT_EQ(0,
              neteq_->GetAudio(
                  kMaxBlockSize, out_data_, &out_len, &num_channels, &type));
    ASSERT_EQ(kBlockSize16kHz, out_len);
    EXPECT_EQ(kOutputCNG, type);
    EXPECT_EQ(timestamp - 10, neteq_->PlayoutTimestamp());
  }

  // Insert speech again.
  ++seq_no;
  timestamp += kCngPeriodSamples;
  PopulateRtpInfo(seq_no, timestamp, &rtp_info);
  ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));

  // Pull audio once and verify that the output is speech again.
  ASSERT_EQ(0,
            neteq_->GetAudio(
                kMaxBlockSize, out_data_, &out_len, &num_channels, &type));
  ASSERT_EQ(kBlockSize16kHz, out_len);
  EXPECT_EQ(kOutputNormal, type);
  EXPECT_EQ(timestamp + kSamples - 10, neteq_->PlayoutTimestamp());
}

TEST_F(NetEqDecodingTest, DiscardDuplicateCng) { DuplicateCng(); }
}  // namespace webrtc