Refactoring to share MockAudioOutputStream implementations across 3 platforms

Re-trying the previous CL. The last change got reverted because of valgrind
failure.

TBR=scherkus

Original CL: http://codereview.chromium.org/155471

Review URL: http://codereview.chromium.org/159167

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@21242 0039d316-1c4b-4281-b951-d872f2087c98

19 files changed, 191 insertions, 191 deletions

diff --git a/media/audio/audio_output.h b/media/audio/audio_output.h
index 4793e71..a12742e 100644
--- a/media/audio/audio_output.h
+++ b/media/audio/audio_output.h
@@ -91,9 +91,7 @@ class AudioOutputStream {
 
   // Starts playing audio and generating AudioSourceCallback::OnMoreData().
   // Since implementor of AudioOutputStream may have internal buffers, right
-  // after calling this method initial buffers are fetched. User of this
-  // object should prepare |AudioOutputStream::GetNumBuffers()| before calling
-  // AudioOutputStream::Start().
+  // after calling this method initial buffers are fetched.
   //
   // The output stream does not take ownership of this callback.
   virtual void Start(AudioSourceCallback* callback) = 0;
@@ -114,11 +112,6 @@ class AudioOutputStream {
   // After calling this method, the object should not be used anymore.
   virtual void Close() = 0;
 
-  // Gets the number of internal buffers used in this output stream. This
-  // method is useful for providing information about how user of this object
-  // should prepare initial buffers before calling AudioOutputStream::Start().
-  virtual size_t GetNumBuffers() = 0;
-
  protected:
   virtual ~AudioOutputStream() {}
 };
@@ -164,11 +157,6 @@ class AudioManager {
   virtual void MuteAll() = 0;
   virtual void UnMuteAll() = 0;
 
-  // For testing purposes only. Returns the internal buffer of the last
-  // AUDIO_MOCK AudioOutputStream closed. Returns NULL if none closed yet.
-  // The buffer size is the same as passed to AudioOutputStream::Open().
-  virtual const void* GetLastMockBuffer() = 0;
-
   // Get AudioManager singleton.
   // TODO(cpu): Define threading requirements for interacting with AudioManager.
   static AudioManager* GetAudioManager();
diff --git a/media/audio/fake_audio_output_stream.cc b/media/audio/fake_audio_output_stream.cc
index e69de29..6eb5417 100644
--- a/media/audio/fake_audio_output_stream.cc
+++ b/media/audio/fake_audio_output_stream.cc
@@ -0,0 +1,72 @@
+// Copyright (c) 2009 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/at_exit.h"
+#include "media/audio/fake_audio_output_stream.h"
+
+bool FakeAudioOutputStream::has_created_fake_stream_ = false;
+FakeAudioOutputStream* FakeAudioOutputStream::last_fake_stream_ = NULL;
+
+// static
+AudioOutputStream* FakeAudioOutputStream::MakeFakeStream() {
+  if (!has_created_fake_stream_)
+    base::AtExitManager::RegisterCallback(&DestroyLastFakeStream, NULL);
+  has_created_fake_stream_ = true;
+
+  return new FakeAudioOutputStream();
+}
+
+// static
+FakeAudioOutputStream* FakeAudioOutputStream::GetLastFakeStream() {
+  return last_fake_stream_;
+}
+
+bool FakeAudioOutputStream::Open(size_t packet_size) {
+  if (packet_size < sizeof(int16))
+    return false;
+  packet_size_ = packet_size;
+  buffer_.reset(new char[packet_size_]);
+  return true;
+}
+
+void FakeAudioOutputStream::Start(AudioSourceCallback* callback)  {
+  callback_ = callback;
+  memset(buffer_.get(), 0, packet_size_);
+  callback_->OnMoreData(this, buffer_.get(), packet_size_);
+}
+
+void FakeAudioOutputStream::Stop() {
+}
+
+void FakeAudioOutputStream::SetVolume(double left_level, double right_level) {
+  left_volume_ = left_level;
+  right_volume_ = right_level;
+}
+
+void FakeAudioOutputStream::GetVolume(double* left_level, double* right_level) {
+  *left_level = left_volume_;
+  *right_level = right_volume_;
+}
+
+void FakeAudioOutputStream::Close() {
+  callback_->OnClose(this);
+  callback_ = NULL;
+
+  if (last_fake_stream_)
+    delete last_fake_stream_;
+  last_fake_stream_ = this;
+}
+
+FakeAudioOutputStream::FakeAudioOutputStream()
+    : left_volume_(0),
+      right_volume_(0),
+      callback_(NULL),
+      packet_size_(0) {
+}
+
+// static
+void FakeAudioOutputStream::DestroyLastFakeStream(void* param) {
+  if (last_fake_stream_)
+    delete last_fake_stream_;
+}
diff --git a/media/audio/fake_audio_output_stream.h b/media/audio/fake_audio_output_stream.h
index e69de29..0cab082 100644
--- a/media/audio/fake_audio_output_stream.h
+++ b/media/audio/fake_audio_output_stream.h
@@ -0,0 +1,50 @@
+// Copyright (c) 2009 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.
+//
+// A fake implementation of AudioOutputStream. It is used for testing purpose.
+// TODO(hclam): Implement a thread in this fake output stream to simulate an
+// audio output stream reading from AudioSourceCallback.
+
+#ifndef MEDIA_AUDIO_FAKE_AUDIO_OUTPUT_STREAM_H_
+#define MEDIA_AUDIO_FAKE_AUDIO_OUTOUT_STREAM_H_
+
+#include <vector>
+
+#include "base/scoped_ptr.h"
+#include "media/audio/audio_output.h"
+
+class FakeAudioOutputStream : public AudioOutputStream {
+ public:
+  static AudioOutputStream* MakeFakeStream();
+  static FakeAudioOutputStream* GetLastFakeStream();
+
+  virtual bool Open(size_t packet_size);
+  virtual void Start(AudioSourceCallback* callback);
+  virtual void Stop();
+  virtual void SetVolume(double left_level, double right_level);
+  virtual void GetVolume(double* left_level, double* right_level);
+  virtual void Close();
+
+  char* buffer() { return buffer_.get(); }
+  double left_volume() { return left_volume_; }
+  double right_volume() { return right_volume_; }
+
+ private:
+  FakeAudioOutputStream();
+  virtual ~FakeAudioOutputStream() {}
+
+  static void DestroyLastFakeStream(void* param);
+  static bool has_created_fake_stream_;
+  static FakeAudioOutputStream* last_fake_stream_;
+
+  double left_volume_;
+  double right_volume_;
+  AudioSourceCallback* callback_;
+  scoped_array<char> buffer_;
+  size_t packet_size_;
+
+  DISALLOW_COPY_AND_ASSIGN(FakeAudioOutputStream);
+};
+
+#endif  // MEDIA_AUDIO_FAKE_AUDIO_OUTPUT_STREAM_H_
diff --git a/media/audio/linux/alsa_output.cc b/media/audio/linux/alsa_output.cc
index e7dab51..08075de 100644
--- a/media/audio/linux/alsa_output.cc
+++ b/media/audio/linux/alsa_output.cc
@@ -527,7 +527,3 @@ void AlsaPCMOutputStream::SetVolume(double left_level, double right_level) {
 void AlsaPCMOutputStream::GetVolume(double* left_level, double* right_level) {
   NOTIMPLEMENTED();
 }
-
-size_t AlsaPCMOutputStream::GetNumBuffers() {
-  return 0;
-}
diff --git a/media/audio/linux/alsa_output.h b/media/audio/linux/alsa_output.h
index 42cd3c8..850ab7f 100644
--- a/media/audio/linux/alsa_output.h
+++ b/media/audio/linux/alsa_output.h
@@ -65,7 +65,6 @@ class AlsaPCMOutputStream :
   virtual void Stop();
   virtual void SetVolume(double left_level, double right_level);
   virtual void GetVolume(double* left_level, double* right_level);
-  virtual size_t GetNumBuffers();
 
  private:
   // Closes the playback handle, reporting errors if any occur.  Returns true
diff --git a/media/audio/linux/audio_manager_linux.cc b/media/audio/linux/audio_manager_linux.cc
index 69e8b9b..a9d299f 100644
--- a/media/audio/linux/audio_manager_linux.cc
+++ b/media/audio/linux/audio_manager_linux.cc
@@ -6,6 +6,7 @@
 
 #include "base/at_exit.h"
 #include "base/logging.h"
+#include "media/audio/fake_audio_output_stream.h"
 #include "media/audio/linux/alsa_output.h"
 
 namespace {
@@ -27,11 +28,15 @@ AudioOutputStream* AudioManagerLinux::MakeAudioStream(Format format,
   // surround40, surround51, etc.
   //
   // http://0pointer.de/blog/projects/guide-to-sound-apis.html
-  AlsaPCMOutputStream* stream =
-      new AlsaPCMOutputStream(AlsaPCMOutputStream::kDefaultDevice,
-                              100 /* 100ms minimal buffer */,
-                              format, channels, sample_rate, bits_per_sample);
-  return stream;
+  if (format == AudioManager::AUDIO_MOCK) {
+    return FakeAudioOutputStream::MakeFakeStream();
+  } else {
+    AlsaPCMOutputStream* stream =
+        new AlsaPCMOutputStream(AlsaPCMOutputStream::kDefaultDevice,
+                                100 /* 100ms minimal buffer */,
+                                format, channels, sample_rate, bits_per_sample);
+    return stream;
+  }
 }
 
 AudioManagerLinux::AudioManagerLinux() {
@@ -50,12 +55,6 @@ void AudioManagerLinux::UnMuteAll() {
   NOTIMPLEMENTED();
 }
 
-const void* AudioManagerLinux::GetLastMockBuffer() {
-  // TODO(ajwong): Implement.
-  NOTIMPLEMENTED();
-  return NULL;
-}
-
 // TODO(ajwong): Collapse this with the windows version.
 void DestroyAudioManagerLinux(void* not_used) {
   delete g_audio_manager;
diff --git a/media/audio/linux/audio_manager_linux.h b/media/audio/linux/audio_manager_linux.h
index 5b369a1..191e687 100644
--- a/media/audio/linux/audio_manager_linux.h
+++ b/media/audio/linux/audio_manager_linux.h
@@ -19,7 +19,6 @@ class AudioManagerLinux : public AudioManager {
                                              char bits_per_sample);
   virtual void MuteAll();
   virtual void UnMuteAll();
-  virtual const void* GetLastMockBuffer();
 
  private:
   // Friend function for invoking the private destructor at exit.
diff --git a/media/audio/mac/audio_manager_mac.cc b/media/audio/mac/audio_manager_mac.cc
index 7a8e15b..a921dfb 100644
--- a/media/audio/mac/audio_manager_mac.cc
+++ b/media/audio/mac/audio_manager_mac.cc
@@ -5,6 +5,7 @@
 #include <CoreAudio/AudioHardware.h>
 
 #include "base/at_exit.h"
+#include "media/audio/fake_audio_output_stream.h"
 #include "media/audio/mac/audio_manager_mac.h"
 #include "media/audio/mac/audio_output_mac.h"
 
@@ -19,8 +20,9 @@ bool AudioManagerMac::HasAudioDevices() {
 AudioOutputStream* AudioManagerMac::MakeAudioStream(Format format, int channels,
                                                     int sample_rate,
                                                     char bits_per_sample) {
-  // TODO(cpu): add mock format.
-  if (format != AUDIO_PCM_LINEAR)
+  if (format == AUDIO_MOCK)
+    return FakeAudioOutputStream::MakeFakeStream();
+  else if (format != AUDIO_PCM_LINEAR)
     return NULL;
   return new PCMQueueOutAudioOutputStream(this, channels, sample_rate,
                                           bits_per_sample);
@@ -34,19 +36,15 @@ void AudioManagerMac::UnMuteAll() {
   // TODO(cpu): implement.
 }
 
-const void* AudioManagerMac::GetLastMockBuffer() {
-  // TODO(cpu): implement.
-  return NULL;
-}
-
 // Called by the stream when it has been released by calling Close().
 void AudioManagerMac::ReleaseStream(PCMQueueOutAudioOutputStream* stream) {
   delete stream;
 }
 
 namespace {
-  AudioManagerMac* g_audio_manager = NULL;
-  
+
+AudioManagerMac* g_audio_manager = NULL;
+
 }  // namespace.
 
 void DestroyAudioManagerMac(void* param) {
diff --git a/media/audio/mac/audio_manager_mac.h b/media/audio/mac/audio_manager_mac.h
index 46cb2a5..51ee58a 100644
--- a/media/audio/mac/audio_manager_mac.h
+++ b/media/audio/mac/audio_manager_mac.h
@@ -24,7 +24,6 @@ class AudioManagerMac : public AudioManager {
                                              char bits_per_sample);
   virtual void MuteAll();
   virtual void UnMuteAll();
-  virtual const void* GetLastMockBuffer();
 
   // Mac-only method to free a stream created in MakeAudioStream.
   // It is called internally by the audio stream when it has been closed.
diff --git a/media/audio/mac/audio_output_mac.cc b/media/audio/mac/audio_output_mac.cc
index 4d7c6bb..657e91a 100644
--- a/media/audio/mac/audio_output_mac.cc
+++ b/media/audio/mac/audio_output_mac.cc
@@ -145,10 +145,6 @@ void PCMQueueOutAudioOutputStream::GetVolume(double* left_level,
   // TODO(cpu): Implement.
 }
 
-size_t PCMQueueOutAudioOutputStream::GetNumBuffers() {
-  return kNumBuffers;
-}
-
 // Note to future hackers of this function: Do not add locks here because we
 // call out to third party source that might do crazy things including adquire
 // external locks or somehow re-enter here because its legal for them to call
diff --git a/media/audio/mac/audio_output_mac.h b/media/audio/mac/audio_output_mac.h
index d1129b3..340e1eb 100644
--- a/media/audio/mac/audio_output_mac.h
+++ b/media/audio/mac/audio_output_mac.h
@@ -35,7 +35,6 @@ class PCMQueueOutAudioOutputStream : public AudioOutputStream {
   virtual void Stop();
   virtual void SetVolume(double left_level, double right_level);
   virtual void GetVolume(double* left_level, double* right_level);
-  virtual size_t GetNumBuffers();
 
  private:
   // The audio is double buffered.
diff --git a/media/audio/simple_sources.cc b/media/audio/simple_sources.cc
index f941782..eaf671d 100644
--- a/media/audio/simple_sources.cc
+++ b/media/audio/simple_sources.cc
@@ -118,7 +118,7 @@ void PushSource::CleanUp() {
   AutoLock auto_lock(lock_);
   PacketList::const_iterator it;
   for (it = packets_.begin(); it != packets_.end(); ++it) {
-    delete it->buffer;
+    delete [] it->buffer;
     buffered_bytes_ -= it->size;
   }
   packets_.clear();
diff --git a/media/audio/simple_sources_unittest.cc b/media/audio/simple_sources_unittest.cc
index 32176b7..ef6a5a7 100644
--- a/media/audio/simple_sources_unittest.cc
+++ b/media/audio/simple_sources_unittest.cc
@@ -6,6 +6,7 @@
 #include "base/basictypes.h"
 #include "base/scoped_ptr.h"
 #include "base/time.h"
+#include "media/audio/fake_audio_output_stream.h"
 #include "media/audio/simple_sources.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
@@ -60,3 +61,48 @@ TEST(SimpleSourcesTest, PushSourceSmallerWrite) {
 
   push_source.OnClose(NULL);
 }
+
+// Validate that the SineWaveAudioSource writes the expected values for
+// the FORMAT_16BIT_MONO. The values are carefully selected so rounding issues
+// do not affect the result. We also test that AudioManager::GetLastMockBuffer
+// works.
+TEST(SimpleSources, SineWaveAudio16MonoTest) {
+  const size_t samples = 1024;
+  const size_t bytes_per_sample = 2;
+  const int freq = 200;
+
+  SineWaveAudioSource source(SineWaveAudioSource::FORMAT_16BIT_LINEAR_PCM, 1,
+                             freq, AudioManager::kTelephoneSampleRate);
+
+  AudioManager* audio_man = AudioManager::GetAudioManager();
+  ASSERT_TRUE(NULL != audio_man);
+  AudioOutputStream* oas =
+      audio_man->MakeAudioStream(AudioManager::AUDIO_MOCK, 1,
+                                 AudioManager::kTelephoneSampleRate,
+                                 bytes_per_sample * 2);
+  ASSERT_TRUE(NULL != oas);
+  EXPECT_TRUE(oas->Open(samples * bytes_per_sample));
+
+  oas->Start(&source);
+  oas->Stop();
+  oas->Close();
+
+  ASSERT_TRUE(FakeAudioOutputStream::GetLastFakeStream());
+  const int16* last_buffer =
+      reinterpret_cast<int16*>(
+          FakeAudioOutputStream::GetLastFakeStream()->buffer());
+  ASSERT_TRUE(NULL != last_buffer);
+
+  size_t half_period = AudioManager::kTelephoneSampleRate / (freq * 2);
+
+  // Spot test positive incursion of sine wave.
+  EXPECT_EQ(0, last_buffer[0]);
+  EXPECT_EQ(5126, last_buffer[1]);
+  EXPECT_TRUE(last_buffer[1] < last_buffer[2]);
+  EXPECT_TRUE(last_buffer[2] < last_buffer[3]);
+  // Spot test negative incursion of sine wave.
+  EXPECT_EQ(0, last_buffer[half_period]);
+  EXPECT_EQ(-5126, last_buffer[half_period + 1]);
+  EXPECT_TRUE(last_buffer[half_period + 1] > last_buffer[half_period + 2]);
+  EXPECT_TRUE(last_buffer[half_period + 2] > last_buffer[half_period + 3]);
+}
diff --git a/media/audio/win/audio_manager_win.h b/media/audio/win/audio_manager_win.h
index 12a4c06..f060fc4 100644
--- a/media/audio/win/audio_manager_win.h
+++ b/media/audio/win/audio_manager_win.h
@@ -11,7 +11,6 @@
 #include "media/audio/audio_output.h"
 
 class PCMWaveOutAudioOutputStream;
-class AudioOutputStreamMockWin;
 
 // Windows implementation of the AudioManager singleton. This class is internal
 // to the audio output and only internal users can call methods not exposed by
@@ -26,12 +25,10 @@ class AudioManagerWin : public AudioManager {
                                              char bits_per_sample);
   virtual void MuteAll();
   virtual void UnMuteAll();
-  virtual const void* GetLastMockBuffer();
 
   // Windows-only methods to free a stream created in MakeAudioStream. These
   // are called internally by the audio stream when it has been closed.
   void ReleaseStream(PCMWaveOutAudioOutputStream* stream);
-  void ReleaseStream(AudioOutputStreamMockWin* stream);
 
  private:
   friend void DestroyAudioManagerWin(void *);
diff --git a/media/audio/win/audio_output_win.cc b/media/audio/win/audio_output_win.cc
index bc8c712..f7eac01 100644
--- a/media/audio/win/audio_output_win.cc
+++ b/media/audio/win/audio_output_win.cc
@@ -9,81 +9,10 @@
 
 #include "base/at_exit.h"
 #include "base/basictypes.h"
+#include "media/audio/fake_audio_output_stream.h"
 #include "media/audio/win/audio_manager_win.h"
 #include "media/audio/win/waveout_output_win.h"
 
-// A do-nothing audio stream. It behaves like a regular audio stream but does
-// not have any side effect, except possibly the creation and tear-down of
-// of a thread. It is useful to test code that uses audio streams such as
-// audio sources.
-class AudioOutputStreamMockWin : public AudioOutputStream {
- public:
-  explicit AudioOutputStreamMockWin(AudioManagerWin* manager)
-      : manager_(manager),
-        callback_(NULL),
-        buffer_(NULL),
-        packet_size_(0),
-        left_volume_(1.0),
-        right_volume_(1.0) {
-  }
-
-  virtual ~AudioOutputStreamMockWin() {
-    delete[] buffer_;
-    packet_size_ = 0;
-  }
-
-  virtual bool Open(size_t packet_size) {
-    if (packet_size < sizeof(int16))
-      return false;
-    packet_size_ = packet_size;
-    buffer_ = new char[packet_size_];
-    return true;
-  }
-
-  virtual void Start(AudioSourceCallback* callback)  {
-    callback_ = callback;
-    memset(buffer_, 0, packet_size_);
-    callback_->OnMoreData(this, buffer_, packet_size_);
-  }
-
-  // TODO(cpu): flesh out Start and Stop methods. We need a thread to
-  // perform periodic callbacks.
-  virtual void Stop() {
-  }
-
-  virtual void SetVolume(double left_level, double right_level) {
-    left_volume_ = left_level;
-    right_volume_ = right_level;
-  }
-
-  virtual void GetVolume(double* left_level, double* right_level) {
-    *left_level = left_volume_;
-    *right_level = right_volume_;
-  }
-
-  virtual size_t GetNumBuffers() {
-    return 1;
-  }
-
-  virtual void Close() {
-    callback_->OnClose(this);
-    callback_ = NULL;
-    manager_->ReleaseStream(this);
-  }
-
-  char* buffer() {
-    return buffer_;
-  }
-
- private:
-  AudioManagerWin* manager_;
-  AudioSourceCallback* callback_;
-  char* buffer_;
-  size_t packet_size_;
-  double left_volume_;
-  double right_volume_;
-};
-
 namespace {
 
 // The next 3 constants are some sensible limits to prevent integer overflow
@@ -102,18 +31,10 @@ const int kMaxChannels = 6;
 const int kMaxSampleRate = 192000;
 const int kMaxBitsPerSample = 64;
 
-AudioOutputStreamMockWin* g_last_mock_stream = NULL;
 AudioManagerWin* g_audio_manager = NULL;
 
-void ReplaceLastMockStream(AudioOutputStreamMockWin* newer) {
-  if (g_last_mock_stream)
-    delete g_last_mock_stream;
-  g_last_mock_stream = newer;
-}
-
 }  // namespace.
 
-
 bool AudioManagerWin::HasAudioDevices() {
   return (::waveOutGetNumDevs() != 0);
 }
@@ -122,7 +43,6 @@ bool AudioManagerWin::HasAudioDevices() {
 // should suffice most windows user's needs.
 // - PCMWaveOutAudioOutputStream: Based on the waveOutWrite API (in progress)
 // - PCMDXSoundAudioOutputStream: Based on DirectSound or XAudio (future work).
-
 AudioOutputStream* AudioManagerWin::MakeAudioStream(Format format, int channels,
                                                     int sample_rate,
                                                     char bits_per_sample) {
@@ -132,7 +52,7 @@ AudioOutputStream* AudioManagerWin::MakeAudioStream(Format format, int channels,
     return NULL;
 
   if (format == AUDIO_MOCK) {
-    return new AudioOutputStreamMockWin(this);
+    return FakeAudioOutputStream::MakeFakeStream();
   } else if (format == AUDIO_PCM_LINEAR) {
     return new PCMWaveOutAudioOutputStream(this, channels, sample_rate,
                                            bits_per_sample, WAVE_MAPPER);
@@ -145,15 +65,6 @@ void AudioManagerWin::ReleaseStream(PCMWaveOutAudioOutputStream* stream) {
     delete stream;
 }
 
-void AudioManagerWin::ReleaseStream(AudioOutputStreamMockWin *stream) {
-  // Note that we keep the last mock stream so GetLastMockBuffer() works.
-  ReplaceLastMockStream(stream);
-}
-
-const void* AudioManagerWin::GetLastMockBuffer() {
-  return (g_last_mock_stream) ? g_last_mock_stream->buffer() : NULL;
-}
-
 void AudioManagerWin::MuteAll() {
 }
 
@@ -161,7 +72,6 @@ void AudioManagerWin::UnMuteAll() {
 }
 
 AudioManagerWin::~AudioManagerWin() {
-  ReplaceLastMockStream(NULL);
 }
 
 void DestroyAudioManagerWin(void* param) {
diff --git a/media/audio/win/audio_output_win_unittest.cc b/media/audio/win/audio_output_win_unittest.cc
index fae74ac..f47bd9c 100644
--- a/media/audio/win/audio_output_win_unittest.cc
+++ b/media/audio/win/audio_output_win_unittest.cc
@@ -193,51 +193,6 @@ TEST(WinAudioTest, MockStreamBasicCallbacks) {
   EXPECT_EQ(1, source.was_closed());
 }
 
-// Validate that the SineWaveAudioSource writes the expected values for
-// the FORMAT_16BIT_MONO. The values are carefully selected so rounding issues
-// do not affect the result. We also test that AudioManager::GetLastMockBuffer
-// works.
-// TODO(hclam): move this test to SimpleSourcesTest once mock audio stream is
-// implemented on other platform.
-TEST(WinAudioTest, SineWaveAudio16MonoTest) {
-  const size_t samples = 1024;
-  const size_t bytes_per_sample = 2;
-  const int freq = 200;
-
-  SineWaveAudioSource source(SineWaveAudioSource::FORMAT_16BIT_LINEAR_PCM, 1,
-                             freq, AudioManager::kTelephoneSampleRate);
-
-  AudioManager* audio_man = AudioManager::GetAudioManager();
-  ASSERT_TRUE(NULL != audio_man);
-  AudioOutputStream* oas =
-      audio_man->MakeAudioStream(AudioManager::AUDIO_MOCK, 1,
-                                 AudioManager::kTelephoneSampleRate,
-                                 bytes_per_sample * 2);
-  ASSERT_TRUE(NULL != oas);
-  EXPECT_TRUE(oas->Open(samples * bytes_per_sample));
-
-  oas->Start(&source);
-  oas->Stop();
-  oas->Close();
-
-  const int16* last_buffer =
-      reinterpret_cast<const int16*>(audio_man->GetLastMockBuffer());
-  ASSERT_TRUE(NULL != last_buffer);
-
-  size_t half_period = AudioManager::kTelephoneSampleRate / (freq * 2);
-
-  // Spot test positive incursion of sine wave.
-  EXPECT_EQ(0, last_buffer[0]);
-  EXPECT_EQ(5126, last_buffer[1]);
-  EXPECT_TRUE(last_buffer[1] < last_buffer[2]);
-  EXPECT_TRUE(last_buffer[2] < last_buffer[3]);
-  // Spot test negative incursion of sine wave.
-  EXPECT_EQ(0, last_buffer[half_period]);
-  EXPECT_EQ(-5126, last_buffer[half_period + 1]);
-  EXPECT_TRUE(last_buffer[half_period + 1] > last_buffer[half_period + 2]);
-  EXPECT_TRUE(last_buffer[half_period + 2] > last_buffer[half_period + 3]);
-}
-
 // ===========================================================================
 // Validation of AudioManager::AUDIO_PCM_LINEAR
 //
diff --git a/media/audio/win/waveout_output_win.cc b/media/audio/win/waveout_output_win.cc
index d8d2ebdd..124d8e7 100644
--- a/media/audio/win/waveout_output_win.cc
+++ b/media/audio/win/waveout_output_win.cc
@@ -215,10 +215,6 @@ void PCMWaveOutAudioOutputStream::GetVolume(double* left_level,
   *right_level = volume_;
 }
 
-size_t PCMWaveOutAudioOutputStream::GetNumBuffers() {
-  return kNumBuffers;
-}
-
 void PCMWaveOutAudioOutputStream::HandleError(MMRESULT error) {
   DLOG(WARNING) << "PCMWaveOutAudio error " << error;
   callback_->OnError(this, error);
diff --git a/media/audio/win/waveout_output_win.h b/media/audio/win/waveout_output_win.h
index ea9bdef..3f822e9 100644
--- a/media/audio/win/waveout_output_win.h
+++ b/media/audio/win/waveout_output_win.h
@@ -40,7 +40,6 @@ class PCMWaveOutAudioOutputStream : public AudioOutputStream {
   virtual void Stop();
   virtual void SetVolume(double left_level, double right_level);
   virtual void GetVolume(double* left_level, double* right_level);
-  virtual size_t GetNumBuffers();
 
   // Sends a buffer to the audio driver for playback.
   void QueueNextPacket(WAVEHDR* buffer);
diff --git a/media/media.gyp b/media/media.gyp
index b9c0fa4..853d3db 100644
--- a/media/media.gyp
+++ b/media/media.gyp
@@ -32,6 +32,8 @@
         'audio/audio_output.h',
         'audio/audio_util.cc',
         'audio/audio_util.h',
+        'audio/fake_audio_output_stream.cc',
+        'audio/fake_audio_output_stream.h',
         'audio/linux/audio_manager_linux.cc',
         'audio/linux/audio_manager_linux.h',
         'audio/linux/alsa_output.cc',