// Copyright 2013 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.

#include <stdint.h>

#include <memory>

#include "base/android/build_info.h"
#include "base/bind.h"
#include "base/files/file_util.h"
#include "base/macros.h"
#include "base/message_loop/message_loop.h"
#include "base/path_service.h"
#include "base/run_loop.h"
#include "base/strings/stringprintf.h"
#include "base/synchronization/lock.h"
#include "base/synchronization/waitable_event.h"
#include "base/test/test_timeouts.h"
#include "base/threading/thread_task_runner_handle.h"
#include "base/time/time.h"
#include "build/build_config.h"
#include "media/audio/android/audio_manager_android.h"
#include "media/audio/audio_device_description.h"
#include "media/audio/audio_device_info_accessor_for_tests.h"
#include "media/audio/audio_io.h"
#include "media/audio/audio_unittest_util.h"
#include "media/audio/mock_audio_source_callback.h"
#include "media/audio/test_audio_thread.h"
#include "media/base/decoder_buffer.h"
#include "media/base/seekable_buffer.h"
#include "media/base/test_data_util.h"
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"

using ::testing::_;
using ::testing::AtLeast;
using ::testing::DoAll;
using ::testing::Invoke;
using ::testing::NotNull;
using ::testing::Return;

namespace media {
namespace {

ACTION_P4(CheckCountAndPostQuitTask, count, limit, task_runner, quit_closure) {
  if (++*count >= limit)
    task_runner->PostTask(FROM_HERE, quit_closure);
}

const char kSpeechFile_16b_s_48k[] = "speech_16b_stereo_48kHz.raw";
const char kSpeechFile_16b_m_48k[] = "speech_16b_mono_48kHz.raw";
const char kSpeechFile_16b_s_44k[] = "speech_16b_stereo_44kHz.raw";
const char kSpeechFile_16b_m_44k[] = "speech_16b_mono_44kHz.raw";

const float kCallbackTestTimeMs = 2000.0;
const int kBytesPerSample = 2;
const SampleFormat kSampleFormat = kSampleFormatS16;

// Converts AudioParameters::Format enumerator to readable string.
std::string FormatToString(AudioParameters::Format format) {
  switch (format) {
    case AudioParameters::AUDIO_PCM_LINEAR:
      return std::string("AUDIO_PCM_LINEAR");
    case AudioParameters::AUDIO_PCM_LOW_LATENCY:
      return std::string("AUDIO_PCM_LOW_LATENCY");
    case AudioParameters::AUDIO_FAKE:
      return std::string("AUDIO_FAKE");
    default:
      return std::string();
  }
}

// Converts ChannelLayout enumerator to readable string. Does not include
// multi-channel cases since these layouts are not supported on Android.
std::string LayoutToString(ChannelLayout channel_layout) {
  switch (channel_layout) {
    case CHANNEL_LAYOUT_NONE:
      return std::string("CHANNEL_LAYOUT_NONE");
    case CHANNEL_LAYOUT_MONO:
      return std::string("CHANNEL_LAYOUT_MONO");
    case CHANNEL_LAYOUT_STEREO:
      return std::string("CHANNEL_LAYOUT_STEREO");
    case CHANNEL_LAYOUT_UNSUPPORTED:
    default:
      return std::string("CHANNEL_LAYOUT_UNSUPPORTED");
  }
}

double ExpectedTimeBetweenCallbacks(AudioParameters params) {
  return (base::TimeDelta::FromMicroseconds(
              params.frames_per_buffer() * base::Time::kMicrosecondsPerSecond /
              static_cast<double>(params.sample_rate()))).InMillisecondsF();
}

// Helper method which verifies that the device list starts with a valid
// default device name followed by non-default device names.
void CheckDeviceDescriptions(
    const AudioDeviceDescriptions& device_descriptions) {
  DVLOG(2) << "Got " << device_descriptions.size() << " audio devices.";
  if (device_descriptions.empty()) {
    // Log a warning so we can see the status on the build bots.  No need to
    // break the test though since this does successfully test the code and
    // some failure cases.
    LOG(WARNING) << "No input devices detected";
    return;
  }

  AudioDeviceDescriptions::const_iterator it = device_descriptions.begin();

  // The first device in the list should always be the default device.
  EXPECT_EQ(std::string(AudioDeviceDescription::kDefaultDeviceId),
            it->unique_id);
  ++it;

  // Other devices should have non-empty name and id and should not contain
  // default name or id.
  while (it != device_descriptions.end()) {
    EXPECT_FALSE(it->device_name.empty());
    EXPECT_FALSE(it->unique_id.empty());
    EXPECT_FALSE(it->group_id.empty());
    DVLOG(2) << "Device ID(" << it->unique_id << "), label: " << it->device_name
             << " group: " << it->group_id;
    EXPECT_NE(AudioDeviceDescription::GetDefaultDeviceName(), it->device_name);
    EXPECT_NE(std::string(AudioDeviceDescription::kDefaultDeviceId),
              it->unique_id);
    ++it;
  }
}

// We clear the data bus to ensure that the test does not cause noise.
int RealOnMoreData(base::TimeDelta /* delay */,
                   base::TimeTicks /* delay_timestamp */,
                   int /* prior_frames_skipped */,
                   AudioBus* dest) {
  dest->Zero();
  return dest->frames();
}

}  // namespace

std::ostream& operator<<(std::ostream& os, const AudioParameters& params) {
  using std::endl;
  os << endl
     << "format: " << FormatToString(params.format()) << endl
     << "channel layout: " << LayoutToString(params.channel_layout()) << endl
     << "sample rate: " << params.sample_rate() << endl
     << "frames per buffer: " << params.frames_per_buffer() << endl
     << "channels: " << params.channels() << endl
     << "bytes per buffer: " << params.GetBytesPerBuffer(kSampleFormat) << endl
     << "bytes per second: "
     << params.sample_rate() * params.GetBytesPerFrame(kSampleFormat) << endl
     << "bytes per frame: " << params.GetBytesPerFrame(kSampleFormat) << endl
     << "chunk size in ms: " << ExpectedTimeBetweenCallbacks(params) << endl
     << "echo_canceller: "
     << (params.effects() & AudioParameters::ECHO_CANCELLER);
  return os;
}

// Gmock implementation of AudioInputStream::AudioInputCallback.
class MockAudioInputCallback : public AudioInputStream::AudioInputCallback {
 public:
  MOCK_METHOD3(OnData,
               void(const AudioBus* src,
                    base::TimeTicks capture_time,
                    double volume));
  MOCK_METHOD0(OnError, void());
};

// Implements AudioOutputStream::AudioSourceCallback and provides audio data
// by reading from a data file.
class FileAudioSource : public AudioOutputStream::AudioSourceCallback {
 public:
  explicit FileAudioSource(base::WaitableEvent* event, const std::string& name)
      : event_(event), pos_(0) {
    // Reads a test file from media/test/data directory and stores it in
    // a DecoderBuffer.
    file_ = ReadTestDataFile(name);

    // Log the name of the file which is used as input for this test.
    base::FilePath file_path = GetTestDataFilePath(name);
    DVLOG(0) << "Reading from file: " << file_path.value().c_str();
  }

  ~FileAudioSource() override {}

  // AudioOutputStream::AudioSourceCallback implementation.

  // Use samples read from a data file and fill up the audio buffer
  // provided to us in the callback.
  int OnMoreData(base::TimeDelta /* delay */,
                 base::TimeTicks /* delay_timestamp */,
                 int /* prior_frames_skipped */,
                 AudioBus* dest) override {
    bool stop_playing = false;
    int max_size = dest->frames() * dest->channels() * kBytesPerSample;

    // Adjust data size and prepare for end signal if file has ended.
    if (pos_ + max_size > file_size()) {
      stop_playing = true;
      max_size = file_size() - pos_;
    }

    // File data is stored as interleaved 16-bit values. Copy data samples from
    // the file and deinterleave to match the audio bus format.
    // FromInterleaved() will zero out any unfilled frames when there is not
    // sufficient data remaining in the file to fill up the complete frame.
    int frames = max_size / (dest->channels() * kBytesPerSample);
    if (max_size) {
      dest->FromInterleaved(file_->data() + pos_, frames, kBytesPerSample);
      pos_ += max_size;
    }

    // Set event to ensure that the test can stop when the file has ended.
    if (stop_playing)
      event_->Signal();

    return frames;
  }

  void OnError() override {}

  int file_size() { return file_->data_size(); }

 private:
  base::WaitableEvent* event_;
  int pos_;
  scoped_refptr<DecoderBuffer> file_;

  DISALLOW_COPY_AND_ASSIGN(FileAudioSource);
};

// Implements AudioInputStream::AudioInputCallback and writes the recorded
// audio data to a local output file. Note that this implementation should
// only be used for manually invoked and evaluated tests, hence the created
// file will not be destroyed after the test is done since the intention is
// that it shall be available for off-line analysis.
class FileAudioSink : public AudioInputStream::AudioInputCallback {
 public:
  explicit FileAudioSink(base::WaitableEvent* event,
                         const AudioParameters& params,
                         const std::string& file_name)
      : event_(event), params_(params) {
    // Allocate space for ~10 seconds of data.
    const int kMaxBufferSize =
        10 * params.sample_rate() * params.GetBytesPerFrame(kSampleFormat);
    buffer_.reset(new media::SeekableBuffer(0, kMaxBufferSize));

    // Open up the binary file which will be written to in the destructor.
    base::FilePath file_path;
    EXPECT_TRUE(base::PathService::Get(base::DIR_SOURCE_ROOT, &file_path));
    file_path = file_path.AppendASCII(file_name.c_str());
    binary_file_ = base::OpenFile(file_path, "wb");
    DLOG_IF(ERROR, !binary_file_) << "Failed to open binary PCM data file.";
    DVLOG(0) << "Writing to file: " << file_path.value().c_str();
  }

  ~FileAudioSink() override {
    int bytes_written = 0;
    while (bytes_written < buffer_->forward_capacity()) {
      const uint8_t* chunk;
      int chunk_size;

      // Stop writing if no more data is available.
      if (!buffer_->GetCurrentChunk(&chunk, &chunk_size))
        break;

      // Write recorded data chunk to the file and prepare for next chunk.
      // TODO(henrika): use file_util:: instead.
      fwrite(chunk, 1, chunk_size, binary_file_);
      buffer_->Seek(chunk_size);
      bytes_written += chunk_size;
    }
    base::CloseFile(binary_file_);
  }

  // AudioInputStream::AudioInputCallback implementation.
  void OnData(const AudioBus* src,
              base::TimeTicks capture_time,
              double volume) override {
    const int num_samples = src->frames() * src->channels();
    std::unique_ptr<int16_t> interleaved(new int16_t[num_samples]);
    const int bytes_per_sample = sizeof(*interleaved);
    src->ToInterleaved(src->frames(), bytes_per_sample, interleaved.get());

    // Store data data in a temporary buffer to avoid making blocking
    // fwrite() calls in the audio callback. The complete buffer will be
    // written to file in the destructor.
    const int size = bytes_per_sample * num_samples;
    if (!buffer_->Append((const uint8_t*)interleaved.get(), size))
      event_->Signal();
  }

  void OnError() override {}

 private:
  base::WaitableEvent* event_;
  AudioParameters params_;
  std::unique_ptr<media::SeekableBuffer> buffer_;
  FILE* binary_file_;

  DISALLOW_COPY_AND_ASSIGN(FileAudioSink);
};

// Implements AudioInputCallback and AudioSourceCallback to support full
// duplex audio where captured samples are played out in loopback after
// reading from a temporary FIFO storage.
class FullDuplexAudioSinkSource
    : public AudioInputStream::AudioInputCallback,
      public AudioOutputStream::AudioSourceCallback {
 public:
  explicit FullDuplexAudioSinkSource(const AudioParameters& params)
      : params_(params),
        previous_time_(base::TimeTicks::Now()),
        started_(false) {
    // Start with a reasonably small FIFO size. It will be increased
    // dynamically during the test if required.
    size_t buffer_size = params.GetBytesPerBuffer(kSampleFormat);
    fifo_.reset(new media::SeekableBuffer(0, 2 * buffer_size));
    buffer_.reset(new uint8_t[buffer_size]);
  }

  ~FullDuplexAudioSinkSource() override {}

  // AudioInputStream::AudioInputCallback implementation
  void OnData(const AudioBus* src,
              base::TimeTicks capture_time,
              double volume) override {
    const base::TimeTicks now_time = base::TimeTicks::Now();
    const int diff = (now_time - previous_time_).InMilliseconds();

    const int num_samples = src->frames() * src->channels();
    std::unique_ptr<int16_t> interleaved(new int16_t[num_samples]);
    const int bytes_per_sample = sizeof(*interleaved);
    src->ToInterleaved(src->frames(), bytes_per_sample, interleaved.get());
    const int size = bytes_per_sample * num_samples;

    base::AutoLock lock(lock_);
    if (diff > 1000) {
      started_ = true;
      previous_time_ = now_time;

      // Log out the extra delay added by the FIFO. This is a best effort
      // estimate. We might be +- 10ms off here.
      int extra_fifo_delay =
          static_cast<int>(BytesToMilliseconds(fifo_->forward_bytes() + size));
      DVLOG(1) << extra_fifo_delay;
    }

    // We add an initial delay of ~1 second before loopback starts to ensure
    // a stable callback sequence and to avoid initial bursts which might add
    // to the extra FIFO delay.
    if (!started_)
      return;

    // Append new data to the FIFO and extend the size if the max capacity
    // was exceeded. Flush the FIFO when extended just in case.
    if (!fifo_->Append((const uint8_t*)interleaved.get(), size)) {
      fifo_->set_forward_capacity(2 * fifo_->forward_capacity());
      fifo_->Clear();
    }
  }

  void OnError() override {}

  // AudioOutputStream::AudioSourceCallback implementation
  int OnMoreData(base::TimeDelta /* delay */,
                 base::TimeTicks /* delay_timestamp */,
                 int /* prior_frames_skipped */,
                 AudioBus* dest) override {
    const int size_in_bytes =
        kBytesPerSample * dest->frames() * dest->channels();
    EXPECT_EQ(size_in_bytes, params_.GetBytesPerBuffer(kSampleFormat));

    base::AutoLock lock(lock_);

    // We add an initial delay of ~1 second before loopback starts to ensure
    // a stable callback sequences and to avoid initial bursts which might add
    // to the extra FIFO delay.
    if (!started_) {
      dest->Zero();
      return dest->frames();
    }

    // Fill up destination with zeros if the FIFO does not contain enough
    // data to fulfill the request.
    if (fifo_->forward_bytes() < size_in_bytes) {
      dest->Zero();
    } else {
      fifo_->Read(buffer_.get(), size_in_bytes);
      dest->FromInterleaved<SignedInt16SampleTypeTraits>(
          reinterpret_cast<int16_t*>(buffer_.get()), dest->frames());
    }

    return dest->frames();
  }

 private:
  // Converts from bytes to milliseconds given number of bytes and existing
  // audio parameters.
  double BytesToMilliseconds(int bytes) const {
    const int frames = bytes / params_.GetBytesPerFrame(kSampleFormat);
    return (base::TimeDelta::FromMicroseconds(
                frames * base::Time::kMicrosecondsPerSecond /
                static_cast<double>(params_.sample_rate()))).InMillisecondsF();
  }

  AudioParameters params_;
  base::TimeTicks previous_time_;
  base::Lock lock_;
  std::unique_ptr<media::SeekableBuffer> fifo_;
  std::unique_ptr<uint8_t[]> buffer_;
  bool started_;

  DISALLOW_COPY_AND_ASSIGN(FullDuplexAudioSinkSource);
};

// Test fixture class for tests which only exercise the output path.
class AudioAndroidOutputTest : public testing::Test {
 public:
  AudioAndroidOutputTest()
      : loop_(new base::MessageLoopForUI()),
        audio_manager_(AudioManager::CreateForTesting(
            std::make_unique<TestAudioThread>())),
        audio_manager_device_info_(audio_manager_.get()),
        audio_output_stream_(NULL) {
    // Flush the message loop to ensure that AudioManager is fully initialized.
    base::RunLoop().RunUntilIdle();
  }

  ~AudioAndroidOutputTest() override {
    audio_manager_->Shutdown();
    base::RunLoop().RunUntilIdle();
  }

 protected:
  AudioManager* audio_manager() { return audio_manager_.get(); }
  AudioDeviceInfoAccessorForTests* audio_manager_device_info() {
    return &audio_manager_device_info_;
  }
  const AudioParameters& audio_output_parameters() {
    return audio_output_parameters_;
  }

  // Synchronously runs the provided callback/closure on the audio thread.
  void RunOnAudioThread(const base::Closure& closure) {
    if (!audio_manager()->GetTaskRunner()->BelongsToCurrentThread()) {
      base::WaitableEvent event(
          base::WaitableEvent::ResetPolicy::AUTOMATIC,
          base::WaitableEvent::InitialState::NOT_SIGNALED);
      audio_manager()->GetTaskRunner()->PostTask(
          FROM_HERE,
          base::Bind(&AudioAndroidOutputTest::RunOnAudioThreadImpl,
                     base::Unretained(this),
                     closure,
                     &event));
      event.Wait();
    } else {
      closure.Run();
    }
  }

  void RunOnAudioThreadImpl(const base::Closure& closure,
                            base::WaitableEvent* event) {
    DCHECK(audio_manager()->GetTaskRunner()->BelongsToCurrentThread());
    closure.Run();
    event->Signal();
  }

  void GetDefaultOutputStreamParametersOnAudioThread() {
    RunOnAudioThread(
        base::Bind(&AudioAndroidOutputTest::GetDefaultOutputStreamParameters,
                   base::Unretained(this)));
  }

  void MakeAudioOutputStreamOnAudioThread(const AudioParameters& params) {
    RunOnAudioThread(
        base::Bind(&AudioAndroidOutputTest::MakeOutputStream,
                   base::Unretained(this),
                   params));
  }

  void OpenAndCloseAudioOutputStreamOnAudioThread() {
    RunOnAudioThread(
        base::Bind(&AudioAndroidOutputTest::OpenAndClose,
                   base::Unretained(this)));
  }

  void OpenAndStartAudioOutputStreamOnAudioThread(
      AudioOutputStream::AudioSourceCallback* source) {
    RunOnAudioThread(
        base::Bind(&AudioAndroidOutputTest::OpenAndStart,
                   base::Unretained(this),
                   source));
  }

  void StopAndCloseAudioOutputStreamOnAudioThread() {
    RunOnAudioThread(
        base::Bind(&AudioAndroidOutputTest::StopAndClose,
                   base::Unretained(this)));
  }

  double AverageTimeBetweenCallbacks(int num_callbacks) const {
    return ((end_time_ - start_time_) / static_cast<double>(num_callbacks - 1))
        .InMillisecondsF();
  }

  void StartOutputStreamCallbacks(const AudioParameters& params) {
    double expected_time_between_callbacks_ms =
        ExpectedTimeBetweenCallbacks(params);
    const int num_callbacks =
        (kCallbackTestTimeMs / expected_time_between_callbacks_ms);
    MakeAudioOutputStreamOnAudioThread(params);

    int count = 0;
    MockAudioSourceCallback source;

    base::RunLoop run_loop;
    EXPECT_CALL(source, OnMoreData(_, _, 0, NotNull()))
        .Times(AtLeast(num_callbacks))
        .WillRepeatedly(
            DoAll(CheckCountAndPostQuitTask(&count, num_callbacks,
                                            base::ThreadTaskRunnerHandle::Get(),
                                            run_loop.QuitWhenIdleClosure()),
                  Invoke(RealOnMoreData)));
    EXPECT_CALL(source, OnError()).Times(0);

    OpenAndStartAudioOutputStreamOnAudioThread(&source);

    start_time_ = base::TimeTicks::Now();
    run_loop.Run();
    end_time_ = base::TimeTicks::Now();

    StopAndCloseAudioOutputStreamOnAudioThread();

    double average_time_between_callbacks_ms =
        AverageTimeBetweenCallbacks(num_callbacks);
    DVLOG(0) << "expected time between callbacks: "
             << expected_time_between_callbacks_ms << " ms";
    DVLOG(0) << "average time between callbacks: "
             << average_time_between_callbacks_ms << " ms";
    EXPECT_GE(average_time_between_callbacks_ms,
              0.70 * expected_time_between_callbacks_ms);
    EXPECT_LE(average_time_between_callbacks_ms,
              1.50 * expected_time_between_callbacks_ms);
  }

  void GetDefaultOutputStreamParameters() {
    DCHECK(audio_manager()->GetTaskRunner()->BelongsToCurrentThread());
    audio_output_parameters_ =
        audio_manager_device_info()->GetDefaultOutputStreamParameters();
    EXPECT_TRUE(audio_output_parameters_.IsValid());
  }

  void MakeOutputStream(const AudioParameters& params) {
    DCHECK(audio_manager()->GetTaskRunner()->BelongsToCurrentThread());
    audio_output_stream_ = audio_manager()->MakeAudioOutputStream(
        params, std::string(), AudioManager::LogCallback());
    EXPECT_TRUE(audio_output_stream_);
  }

  void OpenAndClose() {
    DCHECK(audio_manager()->GetTaskRunner()->BelongsToCurrentThread());
    EXPECT_TRUE(audio_output_stream_->Open());
    audio_output_stream_->Close();
    audio_output_stream_ = NULL;
  }

  void OpenAndStart(AudioOutputStream::AudioSourceCallback* source) {
    DCHECK(audio_manager()->GetTaskRunner()->BelongsToCurrentThread());
    EXPECT_TRUE(audio_output_stream_->Open());
    audio_output_stream_->Start(source);
  }

  void StopAndClose() {
    DCHECK(audio_manager()->GetTaskRunner()->BelongsToCurrentThread());
    audio_output_stream_->Stop();
    audio_output_stream_->Close();
    audio_output_stream_ = NULL;
  }

  std::unique_ptr<base::MessageLoopForUI> loop_;
  std::unique_ptr<AudioManager> audio_manager_;
  AudioDeviceInfoAccessorForTests audio_manager_device_info_;
  AudioParameters audio_output_parameters_;
  AudioOutputStream* audio_output_stream_;
  base::TimeTicks start_time_;
  base::TimeTicks end_time_;

 private:
  DISALLOW_COPY_AND_ASSIGN(AudioAndroidOutputTest);
};

// Test fixture class for tests which exercise the input path, or both input and
// output paths. It is value-parameterized to test against both the Java
// AudioRecord (when true) and native OpenSLES (when false) input paths.
class AudioAndroidInputTest : public AudioAndroidOutputTest,
                              public testing::WithParamInterface<bool> {
 public:
  AudioAndroidInputTest() : audio_input_stream_(NULL) {}

 protected:
  const AudioParameters& audio_input_parameters() {
    return audio_input_parameters_;
  }

  AudioParameters GetInputStreamParameters() {
    GetDefaultInputStreamParametersOnAudioThread();

    AudioParameters params = audio_input_parameters();

    // Only the AudioRecord path supports effects, so we can force it to be
    // selected for the test by requesting one. OpenSLES is used otherwise.
    params.set_effects(GetParam() ? AudioParameters::ECHO_CANCELLER
                                  : AudioParameters::NO_EFFECTS);
    return params;
  }

  void GetDefaultInputStreamParametersOnAudioThread() {
     RunOnAudioThread(
        base::Bind(&AudioAndroidInputTest::GetDefaultInputStreamParameters,
                   base::Unretained(this)));
  }

  void MakeAudioInputStreamOnAudioThread(const AudioParameters& params) {
    RunOnAudioThread(
        base::Bind(&AudioAndroidInputTest::MakeInputStream,
                   base::Unretained(this),
                   params));
  }

  void OpenAndCloseAudioInputStreamOnAudioThread() {
    RunOnAudioThread(
        base::Bind(&AudioAndroidInputTest::OpenAndClose,
                   base::Unretained(this)));
  }

  void OpenAndStartAudioInputStreamOnAudioThread(
      AudioInputStream::AudioInputCallback* sink) {
    RunOnAudioThread(
        base::Bind(&AudioAndroidInputTest::OpenAndStart,
                   base::Unretained(this),
                   sink));
  }

  void StopAndCloseAudioInputStreamOnAudioThread() {
    RunOnAudioThread(
        base::Bind(&AudioAndroidInputTest::StopAndClose,
                   base::Unretained(this)));
  }

  void StartInputStreamCallbacks(const AudioParameters& params) {
    double expected_time_between_callbacks_ms =
        ExpectedTimeBetweenCallbacks(params);
    const int num_callbacks =
        (kCallbackTestTimeMs / expected_time_between_callbacks_ms);

    MakeAudioInputStreamOnAudioThread(params);

    int count = 0;
    MockAudioInputCallback sink;

    base::RunLoop run_loop;
    EXPECT_CALL(sink, OnData(NotNull(), _, _))
        .Times(AtLeast(num_callbacks))
        .WillRepeatedly(CheckCountAndPostQuitTask(
            &count, num_callbacks, base::ThreadTaskRunnerHandle::Get(),
            run_loop.QuitWhenIdleClosure()));
    EXPECT_CALL(sink, OnError()).Times(0);

    OpenAndStartAudioInputStreamOnAudioThread(&sink);

    start_time_ = base::TimeTicks::Now();
    run_loop.Run();
    end_time_ = base::TimeTicks::Now();

    StopAndCloseAudioInputStreamOnAudioThread();

    double average_time_between_callbacks_ms =
        AverageTimeBetweenCallbacks(num_callbacks);
    DVLOG(0) << "expected time between callbacks: "
             << expected_time_between_callbacks_ms << " ms";
    DVLOG(0) << "average time between callbacks: "
             << average_time_between_callbacks_ms << " ms";
    EXPECT_GE(average_time_between_callbacks_ms,
              0.70 * expected_time_between_callbacks_ms);
    EXPECT_LE(average_time_between_callbacks_ms,
              1.30 * expected_time_between_callbacks_ms);
  }

  void GetDefaultInputStreamParameters() {
    DCHECK(audio_manager()->GetTaskRunner()->BelongsToCurrentThread());
    audio_input_parameters_ =
        audio_manager_device_info()->GetInputStreamParameters(
            AudioDeviceDescription::kDefaultDeviceId);
  }

  void MakeInputStream(const AudioParameters& params) {
    DCHECK(audio_manager()->GetTaskRunner()->BelongsToCurrentThread());
    audio_input_stream_ = audio_manager()->MakeAudioInputStream(
        params, AudioDeviceDescription::kDefaultDeviceId,
        AudioManager::LogCallback());
    EXPECT_TRUE(audio_input_stream_);
  }

  void OpenAndClose() {
    DCHECK(audio_manager()->GetTaskRunner()->BelongsToCurrentThread());
    EXPECT_TRUE(audio_input_stream_->Open());
    audio_input_stream_->Close();
    audio_input_stream_ = NULL;
  }

  void OpenAndStart(AudioInputStream::AudioInputCallback* sink) {
    DCHECK(audio_manager()->GetTaskRunner()->BelongsToCurrentThread());
    EXPECT_TRUE(audio_input_stream_->Open());
    audio_input_stream_->Start(sink);
  }

  void StopAndClose() {
    DCHECK(audio_manager()->GetTaskRunner()->BelongsToCurrentThread());
    audio_input_stream_->Stop();
    audio_input_stream_->Close();
    audio_input_stream_ = NULL;
  }

  AudioInputStream* audio_input_stream_;
  AudioParameters audio_input_parameters_;

 private:
  DISALLOW_COPY_AND_ASSIGN(AudioAndroidInputTest);
};

// Get the default audio input parameters and log the result.
TEST_P(AudioAndroidInputTest, GetDefaultInputStreamParameters) {
  // We don't go through AudioAndroidInputTest::GetInputStreamParameters() here
  // so that we can log the real (non-overridden) values of the effects.
  GetDefaultInputStreamParametersOnAudioThread();
  EXPECT_TRUE(audio_input_parameters().IsValid());
  DVLOG(1) << audio_input_parameters();
}

// Get the default audio output parameters and log the result.
TEST_F(AudioAndroidOutputTest, GetDefaultOutputStreamParameters) {
  GetDefaultOutputStreamParametersOnAudioThread();
  DVLOG(1) << audio_output_parameters();
}

// Verify input device enumeration.
TEST_F(AudioAndroidInputTest, GetAudioInputDeviceDescriptions) {
  ABORT_AUDIO_TEST_IF_NOT(audio_manager_device_info()->HasAudioInputDevices());
  AudioDeviceDescriptions devices;
  RunOnAudioThread(base::Bind(
      &AudioDeviceInfoAccessorForTests::GetAudioInputDeviceDescriptions,
      base::Unretained(audio_manager_device_info()), &devices));
  CheckDeviceDescriptions(devices);
}

// Verify output device enumeration.
TEST_F(AudioAndroidOutputTest, GetAudioOutputDeviceDescriptions) {
  ABORT_AUDIO_TEST_IF_NOT(audio_manager_device_info()->HasAudioOutputDevices());
  AudioDeviceDescriptions devices;
  RunOnAudioThread(base::Bind(
      &AudioDeviceInfoAccessorForTests::GetAudioOutputDeviceDescriptions,
      base::Unretained(audio_manager_device_info()), &devices));
  CheckDeviceDescriptions(devices);
}

// Ensure that a default input stream can be created and closed.
TEST_P(AudioAndroidInputTest, CreateAndCloseInputStream) {
  AudioParameters params = GetInputStreamParameters();
  MakeAudioInputStreamOnAudioThread(params);
  RunOnAudioThread(
      base::Bind(&AudioInputStream::Close,
                 base::Unretained(audio_input_stream_)));
}

// Ensure that a default output stream can be created and closed.
// TODO(henrika): should we also verify that this API changes the audio mode
// to communication mode, and calls RegisterHeadsetReceiver, the first time
// it is called?
TEST_F(AudioAndroidOutputTest, CreateAndCloseOutputStream) {
  GetDefaultOutputStreamParametersOnAudioThread();
  MakeAudioOutputStreamOnAudioThread(audio_output_parameters());
  RunOnAudioThread(
      base::Bind(&AudioOutputStream::Close,
                 base::Unretained(audio_output_stream_)));
}

// Ensure that a default input stream can be opened and closed.
TEST_P(AudioAndroidInputTest, OpenAndCloseInputStream) {
  AudioParameters params = GetInputStreamParameters();
  MakeAudioInputStreamOnAudioThread(params);
  OpenAndCloseAudioInputStreamOnAudioThread();
}

// Ensure that a default output stream can be opened and closed.
TEST_F(AudioAndroidOutputTest, OpenAndCloseOutputStream) {
  GetDefaultOutputStreamParametersOnAudioThread();
  MakeAudioOutputStreamOnAudioThread(audio_output_parameters());
  OpenAndCloseAudioOutputStreamOnAudioThread();
}

// Start input streaming using default input parameters and ensure that the
// callback sequence is sane.
// Flaky, see crbug.com/683408.
TEST_P(AudioAndroidInputTest, DISABLED_StartInputStreamCallbacks) {
  AudioParameters native_params = GetInputStreamParameters();
  StartInputStreamCallbacks(native_params);
}

// Start input streaming using non default input parameters and ensure that the
// callback sequence is sane. The only change we make in this test is to select
// a 10ms buffer size instead of the default size.
// Flaky, see crbug.com/683408.
TEST_P(AudioAndroidInputTest,
       DISABLED_StartInputStreamCallbacksNonDefaultParameters) {
  AudioParameters params = GetInputStreamParameters();
  params.set_frames_per_buffer(params.sample_rate() / 100);
  StartInputStreamCallbacks(params);
}

// Start output streaming using default output parameters and ensure that the
// callback sequence is sane.
TEST_F(AudioAndroidOutputTest, StartOutputStreamCallbacks) {
  GetDefaultOutputStreamParametersOnAudioThread();
  StartOutputStreamCallbacks(audio_output_parameters());
}

// Start output streaming using non default output parameters and ensure that
// the callback sequence is sane. The only change we make in this test is to
// select a 10ms buffer size instead of the default size and to open up the
// device in mono.
// TODO(henrika): possibly add support for more variations.
TEST_F(AudioAndroidOutputTest, StartOutputStreamCallbacksNonDefaultParameters) {
  GetDefaultOutputStreamParametersOnAudioThread();
  AudioParameters params(audio_output_parameters().format(),
                         CHANNEL_LAYOUT_MONO,
                         audio_output_parameters().sample_rate(),
                         audio_output_parameters().sample_rate() / 100);
  StartOutputStreamCallbacks(params);
}

// Play out a PCM file segment in real time and allow the user to verify that
// the rendered audio sounds OK.
// NOTE: this test requires user interaction and is not designed to run as an
// automatized test on bots.
TEST_F(AudioAndroidOutputTest, DISABLED_RunOutputStreamWithFileAsSource) {
  GetDefaultOutputStreamParametersOnAudioThread();
  DVLOG(1) << audio_output_parameters();
  MakeAudioOutputStreamOnAudioThread(audio_output_parameters());

  std::string file_name;
  const AudioParameters params = audio_output_parameters();
  if (params.sample_rate() == 48000 && params.channels() == 2) {
    file_name = kSpeechFile_16b_s_48k;
  } else if (params.sample_rate() == 48000 && params.channels() == 1) {
    file_name = kSpeechFile_16b_m_48k;
  } else if (params.sample_rate() == 44100 && params.channels() == 2) {
    file_name = kSpeechFile_16b_s_44k;
  } else if (params.sample_rate() == 44100 && params.channels() == 1) {
    file_name = kSpeechFile_16b_m_44k;
  } else {
    FAIL() << "This test supports 44.1kHz and 48kHz mono/stereo only.";
    return;
  }

  base::WaitableEvent event(base::WaitableEvent::ResetPolicy::AUTOMATIC,
                            base::WaitableEvent::InitialState::NOT_SIGNALED);
  FileAudioSource source(&event, file_name);

  OpenAndStartAudioOutputStreamOnAudioThread(&source);
  DVLOG(0) << ">> Verify that the file is played out correctly...";
  EXPECT_TRUE(event.TimedWait(TestTimeouts::action_max_timeout()));
  StopAndCloseAudioOutputStreamOnAudioThread();
}

// Start input streaming and run it for ten seconds while recording to a
// local audio file.
// NOTE: this test requires user interaction and is not designed to run as an
// automatized test on bots.
TEST_P(AudioAndroidInputTest, DISABLED_RunSimplexInputStreamWithFileAsSink) {
  AudioParameters params = GetInputStreamParameters();
  DVLOG(1) << params;
  MakeAudioInputStreamOnAudioThread(params);

  std::string file_name = base::StringPrintf("out_simplex_%d_%d_%d.pcm",
                                             params.sample_rate(),
                                             params.frames_per_buffer(),
                                             params.channels());

  base::WaitableEvent event(base::WaitableEvent::ResetPolicy::AUTOMATIC,
                            base::WaitableEvent::InitialState::NOT_SIGNALED);
  FileAudioSink sink(&event, params, file_name);

  OpenAndStartAudioInputStreamOnAudioThread(&sink);
  DVLOG(0) << ">> Speak into the microphone to record audio...";
  EXPECT_TRUE(event.TimedWait(TestTimeouts::action_max_timeout()));
  StopAndCloseAudioInputStreamOnAudioThread();
}

// Same test as RunSimplexInputStreamWithFileAsSink but this time output
// streaming is active as well (reads zeros only).
// NOTE: this test requires user interaction and is not designed to run as an
// automatized test on bots.
TEST_P(AudioAndroidInputTest, DISABLED_RunDuplexInputStreamWithFileAsSink) {
  AudioParameters in_params = GetInputStreamParameters();
  DVLOG(1) << in_params;
  MakeAudioInputStreamOnAudioThread(in_params);

  GetDefaultOutputStreamParametersOnAudioThread();
  DVLOG(1) << audio_output_parameters();
  MakeAudioOutputStreamOnAudioThread(audio_output_parameters());

  std::string file_name = base::StringPrintf("out_duplex_%d_%d_%d.pcm",
                                             in_params.sample_rate(),
                                             in_params.frames_per_buffer(),
                                             in_params.channels());

  base::WaitableEvent event(base::WaitableEvent::ResetPolicy::AUTOMATIC,
                            base::WaitableEvent::InitialState::NOT_SIGNALED);
  FileAudioSink sink(&event, in_params, file_name);
  MockAudioSourceCallback source;

  EXPECT_CALL(source, OnMoreData(_, _, 0, NotNull()))
      .WillRepeatedly(Invoke(RealOnMoreData));
  EXPECT_CALL(source, OnError()).Times(0);

  OpenAndStartAudioInputStreamOnAudioThread(&sink);
  OpenAndStartAudioOutputStreamOnAudioThread(&source);
  DVLOG(0) << ">> Speak into the microphone to record audio";
  EXPECT_TRUE(event.TimedWait(TestTimeouts::action_max_timeout()));
  StopAndCloseAudioOutputStreamOnAudioThread();
  StopAndCloseAudioInputStreamOnAudioThread();
}

// Start audio in both directions while feeding captured data into a FIFO so
// it can be read directly (in loopback) by the render side. A small extra
// delay will be added by the FIFO and an estimate of this delay will be
// printed out during the test.
// NOTE: this test requires user interaction and is not designed to run as an
// automatized test on bots.
TEST_P(AudioAndroidInputTest,
       DISABLED_RunSymmetricInputAndOutputStreamsInFullDuplex) {
  // Get native audio parameters for the input side.
  AudioParameters default_input_params = GetInputStreamParameters();

  // Modify the parameters so that both input and output can use the same
  // parameters by selecting 10ms as buffer size. This will also ensure that
  // the output stream will be a mono stream since mono is default for input
  // audio on Android.
  AudioParameters io_params = default_input_params;
  default_input_params.set_frames_per_buffer(io_params.sample_rate() / 100);
  DVLOG(1) << io_params;

  // Create input and output streams using the common audio parameters.
  MakeAudioInputStreamOnAudioThread(io_params);
  MakeAudioOutputStreamOnAudioThread(io_params);

  FullDuplexAudioSinkSource full_duplex(io_params);

  // Start a full duplex audio session and print out estimates of the extra
  // delay we should expect from the FIFO. If real-time delay measurements are
  // performed, the result should be reduced by this extra delay since it is
  // something that has been added by the test.
  OpenAndStartAudioInputStreamOnAudioThread(&full_duplex);
  OpenAndStartAudioOutputStreamOnAudioThread(&full_duplex);
  DVLOG(1) << "HINT: an estimate of the extra FIFO delay will be updated "
           << "once per second during this test.";
  DVLOG(0) << ">> Speak into the mic and listen to the audio in loopback...";
  fflush(stdout);
  base::PlatformThread::Sleep(base::TimeDelta::FromSeconds(20));
  printf("\n");
  StopAndCloseAudioOutputStreamOnAudioThread();
  StopAndCloseAudioInputStreamOnAudioThread();
}

INSTANTIATE_TEST_CASE_P(AudioAndroidInputTest,
                        AudioAndroidInputTest,
                        testing::Bool());

}  // namespace media