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_UNSUPPORTED:

[Jói, 2013/09/11 15:42:58]

Could move this to just before default and fall through to the default
implementation.

[henrika (OOO until Aug 14), 2013/09/12 09:47:05]

Done.

+      return std::string("CHANNEL_LAYOUT_UNSUPPORTED");
+    case CHANNEL_LAYOUT_MONO:
+      return std::string("CHANNEL_LAYOUT_MONO");
+    case CHANNEL_LAYOUT_STEREO:
+      return std::string("CHANNEL_LAYOUT_STEREO");
+    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<float>(params.sample_rate()))).InMillisecondsF();

[Jói, 2013/09/11 15:42:58]

The function returns double, not float, is that intentional?

[henrika (OOO until Aug 14), 2013/09/12 09:47:05]

Good point. Fixed.

+}
+
+std::ostream& operator<<(std::ostream& os, const AudioParameters& params) {
+  os << std::endl << "format: " << FormatToString(params.format()) << std::endl

[Jói, 2013/09/11 15:42:58]

I guess this is an artifact of git cl format, but this would really be less
confusing to read if the std::endl were always at the end or the front, or if
they don't fit that way, always on separate lines.

[henrika (OOO until Aug 14), 2013/09/12 09:47:05]

Added local using namespace std.

+     << "channel layout: " << LayoutToString(params.channel_layout())
+     << std::endl << "sample rate: " << params.sample_rate() << std::endl
+     << "bits per sample: " << params.bits_per_sample() << std::endl
+     << "frames per buffer: " << params.frames_per_buffer() << std::endl
+     << "channels: " << params.channels() << std::endl
+     << "bytes per buffer: " << params.GetBytesPerBuffer() << std::endl
+     << "bytes per second: " << params.GetBytesPerSecond() << std::endl
+     << "bytes per frame: " << params.GetBytesPerFrame() << std::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.

[Jói, 2013/09/11 15:42:58]

Would put a note here saying this is only used for manually invoked and
evaluated tests, and that is why you're writing a file without cleaning it up.

[henrika (OOO until Aug 14), 2013/09/12 09:47:05]

Done.

+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_fio_delay =
+          static_cast<int>(BytesToMilliseconds(fifo_->forward_bytes() + size));
+      DVLOG(1) << extra_fio_delay;

[Jói, 2013/09/11 15:42:58]

extra_fio_delay -> extra_fifo_delay

[henrika (OOO until Aug 14), 2013/09/12 09:47:05]

Done.

+    }
+
+    // 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 mac capacity

[Jói, 2013/09/11 15:42:58]

mac -> max

[henrika (OOO until Aug 14), 2013/09/12 09:47:05]

Done.

+    // was exceeded. Flush the FIFO if is 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<float>(params_.sample_rate()))).InMillisecondsF();

[Jói, 2013/09/11 15:42:58]

Again float vs. double, is it intentional?

[henrika (OOO until Aug 14), 2013/09/12 09:47:05]

Done.

+  }
+
+  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_) / (num_callbacks - 1)).InMillisecondsF();

[Jói, 2013/09/11 15:42:58]

Looks like the calculation occurs in integer land, is that intentional? Or is
there perhaps some serious operator overloading for TimeTicks that I'm missing?

It is then converted to milliseconds float and finally to double for the return
- is that conversion intentional?

[henrika (OOO until Aug 14), 2013/09/12 09:47:05]

Thanks. Please note that InMillisecondsF() does return double. Odd name I must
say.

+  }
+
+  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());
+  LOG(INFO) << params;

[Jói, 2013/09/11 15:42:58]

For all of the tests that are enabled to run without supervision, I would make
all logs VLOGs that can be easily turned on when doing manual inspection of the
tests but don't litter automated test logs.

This is just my preference though - no need to change any of the logging if the
convention in media code is different.

I understand the motivation behind using LOG(INFO) for the tests and related
code that are meant to run under human supervision. Those I have no problem with
using LOG(INFO).

[henrika (OOO until Aug 14), 2013/09/12 09:47:05]

Done.

+}
+
+// Get the default audio output parameters and log the result.
+TEST_F(AudioAndroidTest, GetDefaultOutputStreamParameters) {
+  AudioParameters params = GetDefaultOutputStreamParameters();
+  EXPECT_TRUE(params.IsValid());
+  LOG(INFO) << 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 changed we make in this test is to

[Jói, 2013/09/11 15:42:58]

changed -> change

[henrika (OOO until Aug 14), 2013/09/12 09:47:05]

Done.

+// 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();
+  LOG(INFO) << 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();
+  LOG(INFO) << 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);
+  DLOG(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);
+  LOG(INFO) << 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);
+  DVLOG(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