Adding audio unit tests for Android

media/audio/android/audio_android_unittest.cc

Index: media/audio/android/audio_android_unittest.cc
diff --git a/media/audio/android/audio_android_unittest.cc b/media/audio/android/audio_android_unittest.cc
new file mode 100644
index 0000000000000000000000000000000000000000..2659129c41eb04287d47f33ee012383ec6999f6a
--- /dev/null
+++ b/media/audio/android/audio_android_unittest.cc
@@ -0,0 +1,839 @@
+// Copyright (c) 2012 The Chromium Authors. All rights reserved.

[wjia(left Chromium), 2013/08/28 22:23:16]

2013.

[henrika (OOO until Aug 14), 2013/08/29 14:13:59]

Done.

+// 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/gtest/include/gtest/gtest.h"
+
+namespace media {
+
+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 int kBitsPerSample = 16;
+
+// TODO(henrika): add commens...
+class MockAudioInputOutputCallbacks
+    : public AudioInputStream::AudioInputCallback,
+      public AudioOutputStream::AudioSourceCallback {
+ public:
+  MockAudioInputOutputCallbacks()
+      : input_callbacks_(0),
+        output_callbacks_(0),
+        input_callback_limit_(-1),
+        output_callback_limit_(-1),
+        input_errors_(0),
+        output_errors_(0) {};
+  virtual ~MockAudioInputOutputCallbacks() {};
+
+  // Implementation of AudioInputCallback.
+  virtual void OnData(AudioInputStream* stream, const uint8* src,
+                       uint32 size, uint32 hardware_delay_bytes,
+                       double volume) OVERRIDE {
+    // DVLOG(1) << "+++ OnData +++";
+    // int thread_id = static_cast<int>(base::PlatformThread::CurrentId());
+    // DVLOG(1) << "##" << thread_id;

[wjia(left Chromium), 2013/08/28 22:23:16]

please remove unused code before checking in.

[henrika (OOO until Aug 14), 2013/08/29 14:13:59]

Done.

+
+    if (input_callbacks_ == 0)
+      input_start_time_ = base::TimeTicks::Now();
+
+    input_callbacks_++;
+
+    if (input_callback_limit_ > 0 &&
+        input_callbacks_ == input_callback_limit_) {
+      input_end_time_ = base::TimeTicks::Now();
+      input_event_->Signal();
+    }
+  };
+  virtual void OnClose(AudioInputStream* stream) OVERRIDE {}
+  virtual void OnError(AudioInputStream* stream) OVERRIDE {
+    input_errors_++;
+  }
+
+  // Add comments....
+  virtual int OnMoreData(AudioBus* dest,
+                         AudioBuffersState buffers_state) OVERRIDE {
+    // DVLOG(1) << "--- OnMoreData ---";
+    if (output_callbacks_ == 0)
+      output_start_time_ = base::TimeTicks::Now();
+
+    output_callbacks_++;
+
+    if (output_callback_limit_ > 0 &&
+        output_callbacks_ == output_callback_limit_) {
+      output_end_time_ = base::TimeTicks::Now();
+      output_event_->Signal();
+    }

[wjia(left Chromium), 2013/08/28 22:23:16]

It seems that these lines of code (line 75 through 84) is same as those in
OnData. Any chance to cut dupped code?

[henrika (OOO until Aug 14), 2013/08/29 14:13:59]

I rewrote by creating arrays of size 2, an enumerator for IN and OUT and one
common method. OK?

+
+    dest->Zero();
+    return dest->frames();
+  }
+
+  virtual int OnMoreIOData(AudioBus* source,
+                           AudioBus* dest,
+                           AudioBuffersState buffers_state) {
+    NOTREACHED();
+    return 0;
+  }
+
+  virtual void OnError(AudioOutputStream* stream) OVERRIDE {
+    output_errors_++;
+  }
+
+  int input_callbacks() { return input_callbacks_; }
+  void set_input_callback_limit(base::WaitableEvent* event,
+                                int input_callback_limit) {
+    input_event_ = event;
+    input_callback_limit_ = input_callback_limit;
+  }
+  int input_errors() { return input_errors_; }
+  base::TimeTicks input_start_time() { return input_start_time_; }
+  base::TimeTicks input_end_time() { return input_end_time_; }
+
+  int output_callbacks() { return output_callbacks_; }
+  void set_output_callback_limit(base::WaitableEvent* event,
+                                 int output_callback_limit) {
+    output_event_ = event;
+    output_callback_limit_ = output_callback_limit;
+  }
+  int output_errors() { return output_errors_; }
+  base::TimeTicks output_start_time() { return output_start_time_; }
+  base::TimeTicks output_end_time() { return output_end_time_; }
+
+ private:
+  int input_callbacks_;
+  int output_callbacks_;
+  int input_callback_limit_;
+  int output_callback_limit_;
+  int input_errors_;
+  int output_errors_;
+  base::TimeTicks input_start_time_;
+  base::TimeTicks output_start_time_;
+  base::TimeTicks input_end_time_;
+  base::TimeTicks output_end_time_;
+  base::WaitableEvent* input_event_;
+  base::WaitableEvent* output_event_;
+
+  DISALLOW_COPY_AND_ASSIGN(MockAudioInputOutputCallbacks);
+};
+
+// 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),
+        previous_marker_time_(base::TimeTicks::Now()) {
+        // 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);
+        printf("Reading from file: %s\n", file_path.value().c_str());
+        fflush(stdout);
+      }
+
+  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) {
+    // Add a '.'-marker once every second.
+    const base::TimeTicks now_time = base::TimeTicks::Now();
+    const int diff = (now_time - previous_marker_time_).InMilliseconds();
+    if (diff > 1000) {
+      printf(".");
+      fflush(stdout);
+      previous_marker_time_ = now_time;
+    }

[wjia(left Chromium), 2013/08/28 22:23:16]

This won't work well when tests are run in parallel.

[henrika (OOO until Aug 14), 2013/08/29 14:13:59]

Can you elaborate? Not sure if I understand. 

Note that I will most likely have to add DISABLED_ for all tests that requires
manual interaction (like this one). The idea is that not all of theses tests
should run on bots.

+
+    int max_size =
+      audio_bus->frames() * audio_bus->channels() * kBitsPerSample / 8;

[wjia(left Chromium), 2013/08/28 22:23:16]

nit: indent by 4.

[henrika (OOO until Aug 14), 2013/08/29 14:13:59]

Done.

+
+    bool stop_playing = false;
+
+    // Adjust data size and prepare for end signal if file has ended.
+    if (pos_ + static_cast<int>(max_size) > file_size()) {

[wjia(left Chromium), 2013/08/28 22:23:16]

|max_size| is "int".

[henrika (OOO until Aug 14), 2013/08/29 14:13:59]

Done.

+      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() * kBitsPerSample / 8);
+    if (max_size) {
+      audio_bus->FromInterleaved(
+          file_->data() + pos_, frames, kBitsPerSample / 8);
+      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) {}
+
+  int file_size() { return file_->data_size(); }
+
+ private:
+  base::WaitableEvent* event_;
+  int pos_;
+  scoped_refptr<DecoderBuffer> file_;
+  base::TimeTicks previous_marker_time_;
+
+  DISALLOW_COPY_AND_ASSIGN(FileAudioSource);
+};
+
+// Implements AudioInputStream::AudioInputCallback and writes the recorded
+// audio data to a local output file.
+class FileAudioSink : public AudioInputStream::AudioInputCallback {
+ public:
+  explicit FileAudioSink(base::WaitableEvent* event,
+                         const AudioParameters& params,
+                         const std::string& file_name)
+      : event_(event),
+        params_(params),
+        previous_marker_time_(base::TimeTicks::Now()) {
+    // 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.";
+    printf("Writing to file  : %s ", file_path.value().c_str());
+    printf("of size %d bytes\n", buffer_->forward_capacity());
+    fflush(stdout);
+  }
+
+  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.
+      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) {
+    // Add a '.'-marker once every second.
+    const base::TimeTicks now_time = base::TimeTicks::Now();
+    const int diff = (now_time - previous_marker_time_).InMilliseconds();
+    if (diff > 1000) {
+      printf(".");
+      fflush(stdout);
+      previous_marker_time_ = now_time;
+    }
+
+    // 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) {}
+  virtual void OnError(AudioInputStream* stream) {}
+
+ private:
+  base::WaitableEvent* event_;
+  AudioParameters params_;
+  scoped_ptr<media::SeekableBuffer> buffer_;
+  FILE* binary_file_;
+  base::TimeTicks previous_marker_time_;
+
+  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_marker_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 {
+    // Add a '.'-marker once every second.
+    const base::TimeTicks now_time = base::TimeTicks::Now();
+    const int diff = (now_time - previous_marker_time_).InMilliseconds();
+
+    base::AutoLock lock(lock_);
+    if (diff > 1000) {
+      started_ = true;
+      printf(".");
+      fflush(stdout);
+      previous_marker_time_ = now_time;
+    }
+
+    // We add an inital delay of ~1 second before loopback starts to ensure
+    // a stable callback sequcence and to avoid inital burts which might add
+    // to the extra FIFO delay.
+    if (!started_)
+      return;
+
+    if (!fifo_->Append(src, size)) {
+      fifo_->set_forward_capacity(2 * fifo_->forward_capacity());
+    }
+  }
+
+  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 =
+      (kBitsPerSample / 8) * dest->frames() * dest->channels();

[wjia(left Chromium), 2013/08/28 22:23:16]

nit: indent by 4.

[wjia(left Chromium), 2013/08/28 22:23:16]

Do you need kBitsPerSample here? params_ has bits_per_sample_.

[henrika (OOO until Aug 14), 2013/08/29 14:13:59]

Done.

[henrika (OOO until Aug 14), 2013/08/29 14:13:59]

Done.

+    EXPECT_EQ(size_in_bytes, params_.GetBytesPerBuffer());
+
+    base::AutoLock lock(lock_);
+
+    // We add an inital delay of ~1 second before loopback starts to ensure
+    // a stable callback sequcence and to avoid inital burts which might add
+    // to the extra FIFO delay.
+    if (!started_) {
+      dest->Zero();
+      return dest->frames();
+    }
+
+    // Fill up destionation 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(), kBitsPerSample / 8);

[wjia(left Chromium), 2013/08/28 22:23:16]

ditto for kBitsPerSample.

[henrika (OOO until Aug 14), 2013/08/29 14:13:59]

Done.

+    }
+
+    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();
+  }
+
+  AudioParameters params_;
+  base::TimeTicks previous_marker_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()
+      : audio_manager_(AudioManager::Create()) {}
+
+  virtual ~AudioAndroidTest() {}
+
+  AudioManager* audio_manager() { return audio_manager_.get(); }
+
+  // Convenience method which ensures that we are not running on the build
+  // bots and that at least one valid input and output device can be found.
+  bool CanRunAudioTests() {
+    bool input = audio_manager()->HasAudioInputDevices();
+    bool output = audio_manager()->HasAudioOutputDevices();
+    LOG_IF(WARNING, !input) << "No input device detected.";
+    LOG_IF(WARNING, !output) << "No output device detected.";
+    return input && output;

[wjia(left Chromium), 2013/08/28 22:23:16]

I am not sure if I understand the logic here. This test will be built into a apk
file and installed on an android device attached to the build machine. Do you
mean the android device attached to build bots could have no audio input or
output?

[henrika (OOO until Aug 14), 2013/08/29 14:13:59]

Really good point. Did not think of that. Makes sense.

+  }
+
+  // Converts AudioParameters::Format enumerator to readable string.
+  std::string FormatToString(AudioParameters::Format format) {
+    if (format == AudioParameters::AUDIO_PCM_LINEAR)
+      return std::string("AUDIO_PCM_LINEAR");
+    else if (format == AudioParameters::AUDIO_PCM_LOW_LATENCY)
+      return std::string("AUDIO_PCM_LINEAR");
+    else if (format == AudioParameters::AUDIO_FAKE)
+      return std::string("AUDIO_FAKE");
+    else if (format == AudioParameters::AUDIO_LAST_FORMAT)
+      return std::string("AUDIO_LAST_FORMAT");
+    else
+      return std::string();

[wjia(left Chromium), 2013/08/28 22:23:16]

use switch?

[henrika (OOO until Aug 14), 2013/08/29 14:13:59]

Done.

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

[wjia(left Chromium), 2013/08/28 22:23:16]

ditto.

[henrika (OOO until Aug 14), 2013/08/29 14:13:59]

Done.

+  }
+
+  void PrintAudioParameters(AudioParameters params) {
+    printf("format           : %s\n", FormatToString(params.format()).c_str());
+    printf("channel_layout   : %s\n",
+        ChannelLayoutToString(params.channel_layout()).c_str());
+    printf("sample_rate      : %d\n", params.sample_rate());
+    printf("bits_per_sample  : %d\n", params.bits_per_sample());
+    printf("frames_per_buffer: %d\n", params.frames_per_buffer());
+    printf("channels         : %d\n", params.channels());
+    printf("bytes per buffer : %d\n", params.GetBytesPerBuffer());
+    printf("bytes per second : %d\n", params.GetBytesPerSecond());
+    printf("bytes per frame  : %d\n", params.GetBytesPerFrame());
+  }
+
+  AudioParameters GetDefaultInputStreamParameters() {
+    return audio_manager()->GetInputStreamParameters(
+        AudioManagerBase::kDefaultDeviceId);
+  }
+
+  AudioParameters GetDefaultOutputStreamParameters() {
+    return audio_manager()->GetDefaultOutputStreamParameters();
+  }
+
+  double TimeBetweenCallbacks(AudioParameters params) const {

[wjia(left Chromium), 2013/08/28 22:23:16]

It's better to call this "ExpectedTimeBetweenCallbacks", since it calculates the
elapsed time based on # of samples, not real callbacks.

[henrika (OOO until Aug 14), 2013/08/29 14:13:59]

Done.

+  return (base::TimeDelta::FromMicroseconds(

[wjia(left Chromium), 2013/08/28 22:23:16]

nit: indent.

[henrika (OOO until Aug 14), 2013/08/29 14:13:59]

Done.

+      params.frames_per_buffer() * base::Time::kMicrosecondsPerSecond /
+      static_cast<float>(params.sample_rate()))).InMillisecondsF();
+  }
+
+  #define START_STREAM_AND_WAIT_FOR_EVENT(stream)  \
+    EXPECT_TRUE(stream->Open());  \
+    stream->Start(&io_callbacks_);  \
+    EXPECT_TRUE(event.TimedWait(TestTimeouts::action_timeout()));  \
+    stream->Stop();  \
+    stream->Close()
+
+  void StartInputStreamCallbacks(const AudioParameters& params) {
+    double time_between_callbacks_ms = TimeBetweenCallbacks(params);
+    const int num_callbacks = (1000.0 / time_between_callbacks_ms);
+
+    base::WaitableEvent event(false, false);
+    io_callbacks_.set_input_callback_limit(&event, num_callbacks);
+
+    AudioInputStream* ais = audio_manager()->MakeAudioInputStream(
+        params, AudioManagerBase::kDefaultDeviceId);
+    EXPECT_TRUE(ais);
+    START_STREAM_AND_WAIT_FOR_EVENT(ais);
+
+    EXPECT_EQ(io_callbacks_.input_callbacks(), num_callbacks);
+    EXPECT_EQ(io_callbacks_.input_errors(), 0);
+
+    double actual_time_between_callbacks_ms = (
+        (io_callbacks_.input_end_time() - io_callbacks_.input_start_time()) /
+        (io_callbacks_.input_callbacks() - 1)).InMillisecondsF();
+    printf("time between callbacks: %.2fms\n", time_between_callbacks_ms);
+    printf("actual time between callbacks: %.2fms\n",
+        actual_time_between_callbacks_ms);
+    EXPECT_GE(actual_time_between_callbacks_ms,
+        0.75 * time_between_callbacks_ms);
+    EXPECT_LE(actual_time_between_callbacks_ms,
+        1.25 * time_between_callbacks_ms);

[wjia(left Chromium), 2013/08/28 22:23:16]

Is 25% margin good enough for one second audio input/output? 

[henrika (OOO until Aug 14), 2013/08/29 14:13:59]

Good point. So far so good but I've been close on some devices. Makes sense to
tweak. Extended measurement time.

+  }
+
+  void StartOutputStreamCallbacks(const AudioParameters& params) {
+    double time_between_callbacks_ms = TimeBetweenCallbacks(params);
+    const int num_callbacks = (1000.0 / time_between_callbacks_ms);
+
+    base::WaitableEvent event(false, false);
+    io_callbacks_.set_output_callback_limit(&event, num_callbacks);
+    AudioOutputStream* aos = audio_manager()->MakeAudioOutputStream(
+        params, std::string());
+    EXPECT_TRUE(aos);
+    START_STREAM_AND_WAIT_FOR_EVENT(aos);
+
+    EXPECT_EQ(io_callbacks_.output_callbacks(), num_callbacks);
+    EXPECT_EQ(io_callbacks_.output_errors(), 0);
+
+    double actual_time_between_callbacks_ms = (
+        (io_callbacks_.output_end_time() - io_callbacks_.output_start_time()) /
+        (io_callbacks_.output_callbacks() - 1)).InMillisecondsF();
+    printf("time between callbacks: %.2fms\n", time_between_callbacks_ms);
+    printf("actual time between callbacks: %.2fms\n",
+        actual_time_between_callbacks_ms);
+    EXPECT_GE(actual_time_between_callbacks_ms,
+        0.75 * time_between_callbacks_ms);
+    EXPECT_LE(actual_time_between_callbacks_ms,
+        1.25 * time_between_callbacks_ms);
+  }
+
+ protected:
+  base::MessageLoopForUI message_loop_;
+  scoped_ptr<AudioManager> audio_manager_;
+  MockAudioInputOutputCallbacks io_callbacks_;
+
+  DISALLOW_COPY_AND_ASSIGN(AudioAndroidTest);
+};
+
+// Get the default audio input parameters and log the result.
+TEST_F(AudioAndroidTest, GetInputStreamParameters) {
+  if (!CanRunAudioTests())
+    return;
+  AudioParameters params = GetDefaultInputStreamParameters();
+  EXPECT_TRUE(params.IsValid());
+  PrintAudioParameters(params);
+}
+
+// Get the default audio output parameters and log the result.
+TEST_F(AudioAndroidTest, GetDefaultOutputStreamParameters) {
+  if (!CanRunAudioTests())
+    return;
+  AudioParameters params = GetDefaultOutputStreamParameters();
+  EXPECT_TRUE(params.IsValid());
+  PrintAudioParameters(params);
+}
+
+// Check if low-latency output is supported and log the result as output.
+TEST_F(AudioAndroidTest, IsAudioLowLatencySupported) {
+  if (!CanRunAudioTests())
+    return;
+  AudioManagerAndroid* manager =
+      static_cast<AudioManagerAndroid*>(audio_manager());
+  bool low_latency = manager->IsAudioLowLatencySupported();
+  low_latency ?  printf("Low latency output is supported\n") :
+      printf("Low latency output is *not* supported\n");
+}
+
+// Ensure that a default input stream can be created and closed.
+TEST_F(AudioAndroidTest, CreateAndCloseInputStream) {
+  if (!CanRunAudioTests())
+    return;
+  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) {
+  if (!CanRunAudioTests())
+    return;
+  AudioParameters params = GetDefaultOutputStreamParameters();
+  AudioOutputStream* aos = audio_manager()->MakeAudioOutputStream(
+      params, std::string());
+  EXPECT_TRUE(aos);
+  aos->Close();
+}
+
+// Ensure that a default input stream can be opened and closed.
+TEST_F(AudioAndroidTest, OpenAndCloseInputStream) {
+  if (!CanRunAudioTests())
+    return;
+  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) {
+  if (!CanRunAudioTests())
+    return;
+  AudioParameters params = GetDefaultOutputStreamParameters();
+  AudioOutputStream* aos = audio_manager()->MakeAudioOutputStream(
+      params, 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) {
+  if (!CanRunAudioTests())
+    return;
+  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 vatiations.
+TEST_F(AudioAndroidTest, StartInputStreamCallbacksNonDefaultParameters) {
+  if (!CanRunAudioTests())
+    return;
+  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) {
+ if (!CanRunAudioTests())
+    return;
+  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
+// 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 vatiations.
+TEST_F(AudioAndroidTest, StartOutputStreamCallbacksNonDefaultParameters) {
+  if (!CanRunAudioTests())
+    return;
+  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);
+}
+
+TEST_F(AudioAndroidTest, RunOutputStreamWithFileAsSource) {
+  if (!CanRunAudioTests())
+    return;
+
+  AudioParameters params = GetDefaultOutputStreamParameters();
+  AudioOutputStream* aos = audio_manager()->MakeAudioOutputStream(
+      params, std::string());
+  EXPECT_TRUE(aos);
+
+  PrintAudioParameters(params);
+  fflush(stdout);
+
+  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);
+  printf(">> Verify that file is played out correctly");
+  fflush(stdout);
+  EXPECT_TRUE(event.TimedWait(TestTimeouts::action_max_timeout()));
+  printf("\n");
+  aos->Stop();
+  aos->Close();
+}
+
+// Start input streaming and run it for ten seconds while recording to a
+// local audio file.
+TEST_F(AudioAndroidTest, RunSimplexInputStreamWithFileAsSink) {
+  if (!CanRunAudioTests())
+    return;
+
+  AudioParameters params = GetDefaultInputStreamParameters();
+  AudioInputStream* ais = audio_manager()->MakeAudioInputStream(
+      params, AudioManagerBase::kDefaultDeviceId);
+  EXPECT_TRUE(ais);
+
+  PrintAudioParameters(params);
+  fflush(stdout);
+
+  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);
+  printf(">> Speak into the microphone to record audio");
+  fflush(stdout);
+  EXPECT_TRUE(event.TimedWait(TestTimeouts::action_max_timeout()));
+  printf("\n");
+  ais->Stop();
+  ais->Close();
+}
+
+// Same test as RunSimplexInputStreamWithFileAsSink but this time output
+// streaming is active as well (reads zeros only).
+TEST_F(AudioAndroidTest, RunDuplexInputStreamWithFileAsSink) {
+  if (!CanRunAudioTests())
+    return;
+
+  AudioParameters in_params = GetDefaultInputStreamParameters();
+  AudioInputStream* ais = audio_manager()->MakeAudioInputStream(
+      in_params, AudioManagerBase::kDefaultDeviceId);
+  EXPECT_TRUE(ais);
+
+  PrintAudioParameters(in_params);
+  fflush(stdout);
+
+  AudioParameters out_params =
+      audio_manager()->GetDefaultOutputStreamParameters();
+  AudioOutputStream* aos = audio_manager()->MakeAudioOutputStream(
+      out_params, std::string());
+  EXPECT_TRUE(aos);
+
+  PrintAudioParameters(out_params);
+  fflush(stdout);
+
+  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);
+
+  EXPECT_TRUE(ais->Open());
+  EXPECT_TRUE(aos->Open());
+  ais->Start(&sink);
+  aos->Start(&io_callbacks_);
+  printf(">> Speak into the microphone to record audio");
+  fflush(stdout);
+  EXPECT_TRUE(event.TimedWait(TestTimeouts::action_max_timeout()));
+  printf("\n");
+  aos->Stop();
+  ais->Stop();
+  aos->Close();
+  ais->Close();
+}
+
+TEST_F(AudioAndroidTest, RunInputAndOutputStreamsInFullDuplex) {
+  if (!CanRunAudioTests())
+    return;
+
+  // 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);
+
+  PrintAudioParameters(io_params);
+  fflush(stdout);
+
+  // 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());
+  EXPECT_TRUE(aos);
+
+  FullDuplexAudioSinkSource full_duplex(io_params);
+
+  EXPECT_TRUE(ais->Open());
+  EXPECT_TRUE(aos->Open());
+  ais->Start(&full_duplex);
+  aos->Start(&full_duplex);
+  printf(">> Speak into the microphone and listen to the audio in loopback");
+  fflush(stdout);
+  base::PlatformThread::Sleep(base::TimeDelta::FromSeconds(10));
+  printf("\n");
+  aos->Stop();
+  ais->Stop();
+  aos->Close();
+  ais->Close();
+}
+
+}  // namespace media