Adding audio unit tests for Android

media/audio/android/audio_android_unittest.cc

+// 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 "base/basictypes.h"
+#include "base/file_util.h"
+#include "base/memory/scoped_ptr.h"
+#include "base/message_loop/message_loop.h"
+#include "base/path_service.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/time/time.h"
+#include "build/build_config.h"
+#include "media/audio/android/audio_manager_android.h"
+#include "media/audio/audio_io.h"
+#include "media/audio/audio_manager_base.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 {
+
+ACTION_P3(CheckCountAndPostQuitTask, count, limit, loop) {
+  if (++*count >= limit) {
+    loop->PostTask(FROM_HERE, base::MessageLoop::QuitClosure());
+  }
+}
+
+static const char kSpeechFile_16b_s_48k[] = "speech_16b_stereo_48kHz.raw";
+static const char kSpeechFile_16b_m_48k[] = "speech_16b_mono_48kHz.raw";
+static const char kSpeechFile_16b_s_44k[] = "speech_16b_stereo_44kHz.raw";
+static const char kSpeechFile_16b_m_44k[] = "speech_16b_mono_44kHz.raw";
+
+static const float kCallbackTestTimeMs = 2000.0;
+static const int kBitsPerSample = 16;
+static const int kBytesPerSample = kBitsPerSample / 8;
+
+// Converts AudioParameters::Format enumerator to readable string.
+static 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");
+    case AudioParameters::AUDIO_LAST_FORMAT:
+      return std::string("AUDIO_LAST_FORMAT");
+    default:
+      return std::string();
+  }
+}
+
+// Converts ChannelLayout enumerator to readable string. Does not include
+// multi-channel cases since these layouts are not supported on Android.
+static 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");
+  }
+}
+
+static double ExpectedTimeBetweenCallbacks(AudioParameters params) {
+  return (base::TimeDelta::FromMicroseconds(
+              params.frames_per_buffer() * base::Time::kMicrosecondsPerSecond /
+              static_cast<double>(params.sample_rate()))).InMillisecondsF();
+}
+
+std::ostream& operator<<(std::ostream& os, const AudioParameters& params) {
+  using namespace std;
+  os << endl << "format: " << FormatToString(params.format()) << endl
+     << "channel layout: " << LayoutToString(params.channel_layout()) << endl
+     << "sample rate: " << params.sample_rate() << endl
+     << "bits per sample: " << params.bits_per_sample() << endl
+     << "frames per buffer: " << params.frames_per_buffer() << endl
+     << "channels: " << params.channels() << endl
+     << "bytes per buffer: " << params.GetBytesPerBuffer() << endl
+     << "bytes per second: " << params.GetBytesPerSecond() << endl
+     << "bytes per frame: " << params.GetBytesPerFrame() << endl
+     << "frame size in ms: " << ExpectedTimeBetweenCallbacks(params);
+  return os;
+}
+
+// Gmock implementation of AudioInputStream::AudioInputCallback.
+class MockAudioInputCallback : public AudioInputStream::AudioInputCallback {
+ public:
+  MOCK_METHOD5(OnData,
+               void(AudioInputStream* stream,
+                    const uint8* src,
+                    uint32 size,
+                    uint32 hardware_delay_bytes,
+                    double volume));
+  MOCK_METHOD1(OnClose, void(AudioInputStream* stream));
+  MOCK_METHOD1(OnError, void(AudioInputStream* stream));
+};
+
+// Gmock implementation of AudioOutputStream::AudioSourceCallback.
+class MockAudioOutputCallback : public AudioOutputStream::AudioSourceCallback {
+ public:
+  MOCK_METHOD2(OnMoreData,
+               int(AudioBus* dest, AudioBuffersState buffers_state));
+  MOCK_METHOD3(OnMoreIOData,
+               int(AudioBus* source,
+                   AudioBus* dest,
+                   AudioBuffersState buffers_state));
+  MOCK_METHOD1(OnError, void(AudioOutputStream* stream));
+
+  // We clear the data bus to ensure that the test does not cause noise.
+  int RealOnMoreData(AudioBus* dest, AudioBuffersState buffers_state) {
+    dest->Zero();
+    return dest->frames();
+  }
+};
+
+// 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);
+    LOG(INFO) << "Reading from file: " << file_path.value().c_str();
+  }
+
+  virtual ~FileAudioSource() {}
+
+  // AudioOutputStream::AudioSourceCallback implementation.
+
+  // Use samples read from a data file and fill up the audio buffer
+  // provided to us in the callback.
+  virtual int OnMoreData(AudioBus* audio_bus,
+                         AudioBuffersState buffers_state) OVERRIDE {
+    bool stop_playing = false;
+    int max_size =
+        audio_bus->frames() * audio_bus->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 / (audio_bus->channels() * kBytesPerSample);
+    if (max_size) {
+      audio_bus->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;
+  }
+
+  virtual int OnMoreIOData(AudioBus* source,
+                           AudioBus* dest,
+                           AudioBuffersState buffers_state) OVERRIDE {
+    NOTREACHED();
+    return 0;
+  }
+
+  virtual void OnError(AudioOutputStream* stream) 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

[Jói, 2013/09/12 09:50:53]

created -> created file

[henrika (OOO until Aug 14), 2013/09/12 10:02:07]

Done.

+// will not be destroyed after the test is done since the intention that it

[Jói, 2013/09/12 09:50:53]

intention that -> intention is that

[henrika (OOO until Aug 14), 2013/09/12 10:02:07]

Done.

+// 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.GetBytesPerSecond();
+    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(PathService::Get(base::DIR_SOURCE_ROOT, &file_path));
+    file_path = file_path.AppendASCII(file_name.c_str());
+    binary_file_ = file_util::OpenFile(file_path, "wb");
+    DLOG_IF(ERROR, !binary_file_) << "Failed to open binary PCM data file.";
+    LOG(INFO) << "Writing to file: " << file_path.value().c_str();
+  }
+
+  virtual ~FileAudioSink() {
+    int bytes_written = 0;
+    while (bytes_written < buffer_->forward_capacity()) {
+      const uint8* 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;
+    }
+    file_util::CloseFile(binary_file_);
+  }
+
+  // AudioInputStream::AudioInputCallback implementation.
+  virtual void OnData(AudioInputStream* stream,
+                      const uint8* src,
+                      uint32 size,
+                      uint32 hardware_delay_bytes,
+                      double volume) OVERRIDE {
+    // 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.
+    if (!buffer_->Append(src, size))
+      event_->Signal();
+  }
+
+  virtual void OnClose(AudioInputStream* stream) OVERRIDE {}
+  virtual void OnError(AudioInputStream* stream) OVERRIDE {}
+
+ private:
+  base::WaitableEvent* event_;
+  AudioParameters params_;
+  scoped_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.
+    fifo_.reset(new media::SeekableBuffer(0, 2 * params.GetBytesPerBuffer()));
+    buffer_.reset(new uint8[params_.GetBytesPerBuffer()]);
+  }
+
+  virtual ~FullDuplexAudioSinkSource() {}
+
+  // AudioInputStream::AudioInputCallback implementation
+  virtual void OnData(AudioInputStream* stream,
+                      const uint8* src,
+                      uint32 size,
+                      uint32 hardware_delay_bytes,
+                      double volume) OVERRIDE {
+    const base::TimeTicks now_time = base::TimeTicks::Now();
+    const int diff = (now_time - previous_time_).InMilliseconds();
+
+    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(src, size)) {
+      fifo_->set_forward_capacity(2 * fifo_->forward_capacity());
+      fifo_->Clear();
+    }
+  }
+
+  virtual void OnClose(AudioInputStream* stream) OVERRIDE {}
+  virtual void OnError(AudioInputStream* stream) OVERRIDE {}
+
+  // AudioOutputStream::AudioSourceCallback implementation
+  virtual int OnMoreData(AudioBus* dest,
+                         AudioBuffersState buffers_state) OVERRIDE {
+    const int size_in_bytes =
+        (params_.bits_per_sample() / 8) * dest->frames() * dest->channels();
+    EXPECT_EQ(size_in_bytes, params_.GetBytesPerBuffer());
+
+    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(
+          buffer_.get(), dest->frames(), params_.bits_per_sample() / 8);
+    }
+
+    return dest->frames();
+  }
+
+  virtual int OnMoreIOData(AudioBus* source,
+                           AudioBus* dest,
+                           AudioBuffersState buffers_state) OVERRIDE {
+    NOTREACHED();
+    return 0;
+  }
+
+  virtual void OnError(AudioOutputStream* stream) OVERRIDE {}
+
+ 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();
+    return (base::TimeDelta::FromMicroseconds(
+                frames * base::Time::kMicrosecondsPerSecond /
+                static_cast<double>(params_.sample_rate()))).InMillisecondsF();
+  }
+
+  AudioParameters params_;
+  base::TimeTicks previous_time_;
+  base::Lock lock_;
+  scoped_ptr<media::SeekableBuffer> fifo_;
+  scoped_ptr<uint8[]> buffer_;
+  bool started_;
+
+  DISALLOW_COPY_AND_ASSIGN(FullDuplexAudioSinkSource);
+};
+
+// Test fixture class.
+class AudioAndroidTest : public testing::Test {
+ public:
+  AudioAndroidTest() {}
+
+ protected:
+  virtual void SetUp() {
+    audio_manager_.reset(AudioManager::Create());
+    loop_.reset(new base::MessageLoopForUI());
+  }
+
+  virtual void TearDown() {}
+
+  AudioManager* audio_manager() { return audio_manager_.get(); }
+  base::MessageLoopForUI* loop() { return loop_.get(); }
+
+  AudioParameters GetDefaultInputStreamParameters() {
+    return audio_manager()->GetInputStreamParameters(
+        AudioManagerBase::kDefaultDeviceId);
+  }
+
+  AudioParameters GetDefaultOutputStreamParameters() {
+    return audio_manager()->GetDefaultOutputStreamParameters();
+  }
+
+  double AverageTimeBetweenCallbacks(int num_callbacks) const {
+    return ((end_time_ - start_time_) / static_cast<double>(num_callbacks - 1))
+        .InMillisecondsF();
+  }
+
+  void StartInputStreamCallbacks(const AudioParameters& params) {
+    double expected_time_between_callbacks_ms =
+        ExpectedTimeBetweenCallbacks(params);
+    const int num_callbacks =
+        (kCallbackTestTimeMs / expected_time_between_callbacks_ms);
+    AudioInputStream* stream = audio_manager()->MakeAudioInputStream(
+        params, AudioManagerBase::kDefaultDeviceId);
+    EXPECT_TRUE(stream);
+
+    int count = 0;
+    MockAudioInputCallback sink;
+
+    EXPECT_CALL(sink,
+                OnData(stream, NotNull(), params.GetBytesPerBuffer(), _, _))
+        .Times(AtLeast(num_callbacks))
+        .WillRepeatedly(
+             CheckCountAndPostQuitTask(&count, num_callbacks, loop()));
+    EXPECT_CALL(sink, OnError(stream)).Times(0);
+    EXPECT_CALL(sink, OnClose(stream)).Times(1);
+
+    EXPECT_TRUE(stream->Open());
+    stream->Start(&sink);
+    start_time_ = base::TimeTicks::Now();
+    loop()->Run();
+    end_time_ = base::TimeTicks::Now();
+    stream->Stop();
+    stream->Close();
+
+    double average_time_between_callbacks_ms =
+        AverageTimeBetweenCallbacks(num_callbacks);
+    LOG(INFO) << "expected time between callbacks: "
+              << expected_time_between_callbacks_ms << " ms";
+    LOG(INFO) << "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 StartOutputStreamCallbacks(const AudioParameters& params) {
+    double expected_time_between_callbacks_ms =
+        ExpectedTimeBetweenCallbacks(params);
+    const int num_callbacks =
+        (kCallbackTestTimeMs / expected_time_between_callbacks_ms);
+    AudioOutputStream* stream = audio_manager()->MakeAudioOutputStream(
+        params, std::string(), std::string());
+    EXPECT_TRUE(stream);
+
+    int count = 0;
+    MockAudioOutputCallback source;
+
+    EXPECT_CALL(source, OnMoreData(NotNull(), _))
+        .Times(AtLeast(num_callbacks))
+        .WillRepeatedly(
+             DoAll(CheckCountAndPostQuitTask(&count, num_callbacks, loop()),
+                   Invoke(&source, &MockAudioOutputCallback::RealOnMoreData)));
+    EXPECT_CALL(source, OnError(stream)).Times(0);
+    EXPECT_CALL(source, OnMoreIOData(_, _, _)).Times(0);
+
+    EXPECT_TRUE(stream->Open());
+    stream->Start(&source);
+    start_time_ = base::TimeTicks::Now();
+    loop()->Run();
+    end_time_ = base::TimeTicks::Now();
+    stream->Stop();
+    stream->Close();
+
+    double average_time_between_callbacks_ms =
+        AverageTimeBetweenCallbacks(num_callbacks);
+    LOG(INFO) << "expected time between callbacks: "
+              << expected_time_between_callbacks_ms << " ms";
+    LOG(INFO) << "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);
+  }
+
+  scoped_ptr<base::MessageLoopForUI> loop_;
+  scoped_ptr<AudioManager> audio_manager_;
+  base::TimeTicks start_time_;
+  base::TimeTicks end_time_;
+
+  DISALLOW_COPY_AND_ASSIGN(AudioAndroidTest);
+};
+
+// Get the default audio input parameters and log the result.
+TEST_F(AudioAndroidTest, GetInputStreamParameters) {
+  AudioParameters params = GetDefaultInputStreamParameters();
+  EXPECT_TRUE(params.IsValid());
+  VLOG(1) << params;
+}
+
+// Get the default audio output parameters and log the result.
+TEST_F(AudioAndroidTest, GetDefaultOutputStreamParameters) {
+  AudioParameters params = GetDefaultOutputStreamParameters();
+  EXPECT_TRUE(params.IsValid());
+  VLOG(1) << params;
+}
+
+// Check if low-latency output is supported and log the result as output.
+TEST_F(AudioAndroidTest, IsAudioLowLatencySupported) {
+  AudioManagerAndroid* manager =
+      static_cast<AudioManagerAndroid*>(audio_manager());
+  bool low_latency = manager->IsAudioLowLatencySupported();
+  low_latency ? LOG(INFO) << "Low latency output is supported"
+              : LOG(INFO) << "Low latency output is *not* supported";
+}
+
+// Ensure that a default input stream can be created and closed.
+TEST_F(AudioAndroidTest, CreateAndCloseInputStream) {
+  AudioParameters params = GetDefaultInputStreamParameters();
+  AudioInputStream* ais = audio_manager()->MakeAudioInputStream(
+      params, AudioManagerBase::kDefaultDeviceId);
+  EXPECT_TRUE(ais);
+  ais->Close();
+}
+
+// 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(AudioAndroidTest, CreateAndCloseOutputStream) {
+  AudioParameters params = GetDefaultOutputStreamParameters();
+  AudioOutputStream* aos = audio_manager()->MakeAudioOutputStream(
+      params, std::string(), std::string());
+  EXPECT_TRUE(aos);
+  aos->Close();
+}
+
+// Ensure that a default input stream can be opened and closed.
+TEST_F(AudioAndroidTest, OpenAndCloseInputStream) {
+  AudioParameters params = GetDefaultInputStreamParameters();
+  AudioInputStream* ais = audio_manager()->MakeAudioInputStream(
+      params, AudioManagerBase::kDefaultDeviceId);
+  EXPECT_TRUE(ais);
+  EXPECT_TRUE(ais->Open());
+  ais->Close();
+}
+
+// Ensure that a default output stream can be opened and closed.
+TEST_F(AudioAndroidTest, OpenAndCloseOutputStream) {
+  AudioParameters params = GetDefaultOutputStreamParameters();
+  AudioOutputStream* aos = audio_manager()->MakeAudioOutputStream(
+      params, std::string(), std::string());
+  EXPECT_TRUE(aos);
+  EXPECT_TRUE(aos->Open());
+  aos->Close();
+}
+
+// Start input streaming using default input parameters and ensure that the
+// callback sequence is sane.
+TEST_F(AudioAndroidTest, StartInputStreamCallbacks) {
+  AudioParameters params = GetDefaultInputStreamParameters();
+  StartInputStreamCallbacks(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.
+// TODO(henrika): possibly add support for more variations.
+TEST_F(AudioAndroidTest, StartInputStreamCallbacksNonDefaultParameters) {
+  AudioParameters native_params = GetDefaultInputStreamParameters();
+  AudioParameters params(native_params.format(),
+                         native_params.channel_layout(),
+                         native_params.sample_rate(),
+                         native_params.bits_per_sample(),
+                         native_params.sample_rate() / 100);
+  StartInputStreamCallbacks(params);
+}
+
+// Start output streaming using default output parameters and ensure that the
+// callback sequence is sane.
+TEST_F(AudioAndroidTest, StartOutputStreamCallbacks) {
+  AudioParameters params = GetDefaultOutputStreamParameters();
+  StartOutputStreamCallbacks(params);
+}
+
+// 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(AudioAndroidTest, StartOutputStreamCallbacksNonDefaultParameters) {
+  AudioParameters native_params = GetDefaultOutputStreamParameters();
+  AudioParameters params(native_params.format(),
+                         CHANNEL_LAYOUT_MONO,
+                         native_params.sample_rate(),
+                         native_params.bits_per_sample(),
+                         native_params.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(AudioAndroidTest, DISABLED_RunOutputStreamWithFileAsSource) {
+  AudioParameters params = GetDefaultOutputStreamParameters();
+  VLOG(1) << params;
+  AudioOutputStream* aos = audio_manager()->MakeAudioOutputStream(
+      params, std::string(), std::string());
+  EXPECT_TRUE(aos);
+
+  std::string file_name;
+  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(false, false);
+  FileAudioSource source(&event, file_name);
+
+  EXPECT_TRUE(aos->Open());
+  aos->SetVolume(1.0);
+  aos->Start(&source);
+  LOG(INFO) << ">> Verify that the file is played out correctly...";
+  EXPECT_TRUE(event.TimedWait(TestTimeouts::action_max_timeout()));
+  aos->Stop();
+  aos->Close();
+}
+
+// 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_F(AudioAndroidTest, DISABLED_RunSimplexInputStreamWithFileAsSink) {
+  AudioParameters params = GetDefaultInputStreamParameters();
+  VLOG(1) << params;
+  AudioInputStream* ais = audio_manager()->MakeAudioInputStream(
+      params, AudioManagerBase::kDefaultDeviceId);
+  EXPECT_TRUE(ais);
+
+  std::string file_name = base::StringPrintf("out_simplex_%d_%d_%d.pcm",
+                                             params.sample_rate(),
+                                             params.frames_per_buffer(),
+                                             params.channels());
+
+  base::WaitableEvent event(false, false);
+  FileAudioSink sink(&event, params, file_name);
+
+  EXPECT_TRUE(ais->Open());
+  ais->Start(&sink);
+  LOG(INFO) << ">> Speak into the microphone to record audio...";
+  EXPECT_TRUE(event.TimedWait(TestTimeouts::action_max_timeout()));
+  ais->Stop();
+  ais->Close();
+}
+
+// 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_F(AudioAndroidTest, DISABLED_RunDuplexInputStreamWithFileAsSink) {
+  AudioParameters in_params = GetDefaultInputStreamParameters();
+  AudioInputStream* ais = audio_manager()->MakeAudioInputStream(
+      in_params, AudioManagerBase::kDefaultDeviceId);
+  EXPECT_TRUE(ais);
+
+  AudioParameters out_params =
+      audio_manager()->GetDefaultOutputStreamParameters();
+  AudioOutputStream* aos = audio_manager()->MakeAudioOutputStream(
+      out_params, std::string(), std::string());
+  EXPECT_TRUE(aos);
+
+  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(false, false);
+  FileAudioSink sink(&event, in_params, file_name);
+  MockAudioOutputCallback source;
+
+  EXPECT_CALL(source, OnMoreData(NotNull(), _)).WillRepeatedly(
+      Invoke(&source, &MockAudioOutputCallback::RealOnMoreData));
+  EXPECT_CALL(source, OnError(aos)).Times(0);
+  EXPECT_CALL(source, OnMoreIOData(_, _, _)).Times(0);
+
+  EXPECT_TRUE(ais->Open());
+  EXPECT_TRUE(aos->Open());
+  ais->Start(&sink);
+  aos->Start(&source);
+  LOG(INFO) << ">> Speak into the microphone to record audio";
+  EXPECT_TRUE(event.TimedWait(TestTimeouts::action_max_timeout()));
+  aos->Stop();
+  ais->Stop();
+  aos->Close();
+  ais->Close();
+}
+
+// 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_F(AudioAndroidTest,
+       DISABLED_RunSymmetricInputAndOutputStreamsInFullDuplex) {
+  // Get native audio parameters for the input side.
+  AudioParameters default_input_params = GetDefaultInputStreamParameters();
+
+  // 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.format(),
+                            default_input_params.channel_layout(),
+                            default_input_params.sample_rate(),
+                            default_input_params.bits_per_sample(),
+                            default_input_params.sample_rate() / 100);
+  VLOG(1) << io_params;
+
+  // Create input and output streams using the common audio parameters.
+  AudioInputStream* ais = audio_manager()->MakeAudioInputStream(
+      io_params, AudioManagerBase::kDefaultDeviceId);
+  EXPECT_TRUE(ais);
+  AudioOutputStream* aos = audio_manager()->MakeAudioOutputStream(
+      io_params, std::string(), std::string());
+  EXPECT_TRUE(aos);
+
+  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.
+  EXPECT_TRUE(ais->Open());
+  EXPECT_TRUE(aos->Open());
+  ais->Start(&full_duplex);
+  aos->Start(&full_duplex);
+  VLOG(1) << "HINT: an estimate of the extra FIFO delay will be updated "
+          << "once per second during this test.";
+  LOG(INFO) << ">> Speak into the mic and listen to the audio in loopback...";
+  fflush(stdout);
+  base::PlatformThread::Sleep(base::TimeDelta::FromSeconds(20));
+  printf("\n");
+  aos->Stop();
+  ais->Stop();
+  aos->Close();
+  ais->Close();
+}
+
+}  // namespace media