Remove the AudioManager singleton.

media/audio/win/audio_low_latency_output_win_unittest.cc

 // Copyright (c) 2011 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 <windows.h>
 #include <mmsystem.h>
 
 #include "base/basictypes.h"
 #include "base/environment.h"
 #include "base/file_util.h"
@@ skipping 124 matching lines @@
   scoped_array<int> delta_times_;
   int file_size_;
   int pos_;
   base::Time previous_call_time_;
   FILE* text_file_;
   size_t elements_to_write_;
 };
 
 // Convenience method which ensures that we are not running on the build
 // bots and that at least one valid output device can be found.
-static bool CanRunAudioTests() {
+static bool CanRunAudioTests(AudioManager* audio_man) {
+  if (NULL == audio_man)
+    return false;
+
   scoped_ptr<base::Environment> env(base::Environment::Create());
   if (env->HasVar("CHROME_HEADLESS"))
     return false;
-  AudioManager* audio_man = AudioManager::GetAudioManager();
-  if (NULL == audio_man)
-    return false;
   // TODO(henrika): note that we use Wave today to query the number of
   // existing output devices.
   return audio_man->HasAudioOutputDevices();
 }
 
 // Convenience method which creates a default AudioOutputStream object but
 // also allows the user to modify the default settings.
 class AudioOutputStreamWrapper {
  public:
-  AudioOutputStreamWrapper()
+  explicit AudioOutputStreamWrapper(AudioManager* audio_manager)
       : com_init_(ScopedCOMInitializer::kMTA),
-        audio_man_(AudioManager::GetAudioManager()),
+        audio_man_(audio_manager),
         format_(AudioParameters::AUDIO_PCM_LOW_LATENCY),
         channel_layout_(CHANNEL_LAYOUT_STEREO),
         bits_per_sample_(16) {
     // Use native/mixing sample rate and 10ms frame size as default.
     sample_rate_ = static_cast<int>(
         WASAPIAudioOutputStream::HardwareSampleRate(eConsole));
     samples_per_packet_ = sample_rate_ / 100;
     DCHECK(sample_rate_);
   }
 
@@ skipping 27 matching lines @@
  private:
   AudioOutputStream* CreateOutputStream() {
     AudioOutputStream* aos = audio_man_->MakeAudioOutputStream(
         AudioParameters(format_, channel_layout_, sample_rate_,
                         bits_per_sample_, samples_per_packet_));
     EXPECT_TRUE(aos);
     return aos;
   }
 
   ScopedCOMInitializer com_init_;
-  AudioManager* audio_man_;
+  scoped_refptr<AudioManager> audio_man_;
   AudioParameters::Format format_;
   ChannelLayout channel_layout_;
   int bits_per_sample_;
   int sample_rate_;
   int samples_per_packet_;
 };
 
 // Convenience method which creates a default AudioOutputStream object.
-static AudioOutputStream* CreateDefaultAudioOutputStream() {
-  AudioOutputStreamWrapper aosw;
+static AudioOutputStream* CreateDefaultAudioOutputStream(
+    AudioManager* audio_manager) {
+  AudioOutputStreamWrapper aosw(audio_manager);
   AudioOutputStream* aos = aosw.Create();
   return aos;
 }
 
 static void QuitMessageLoop(base::MessageLoopProxy* proxy) {
   proxy->PostTask(FROM_HERE, new MessageLoop::QuitTask());
 }
 
 // Verify that we can retrieve the current hardware/mixing sample rate
 // for all supported device roles. The ERole enumeration defines constants
 // that indicate the role that the system/user has assigned to an audio
 // endpoint device.
 // TODO(henrika): modify this test when we support full device enumeration.
 TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestHardwareSampleRate) {
-  if (!CanRunAudioTests())
+  scoped_refptr<AudioManager> audio_manager(AudioManager::Create());
+  if (!CanRunAudioTests(audio_manager.get()))
     return;
 
   ScopedCOMInitializer com_init(ScopedCOMInitializer::kMTA);
 
   // Default device intended for games, system notification sounds,
   // and voice commands.
   int fs = static_cast<int>(
       WASAPIAudioOutputStream::HardwareSampleRate(eConsole));
   EXPECT_GE(fs, 0);
 
   // Default communication device intended for e.g. VoIP communication.
   fs = static_cast<int>(
       WASAPIAudioOutputStream::HardwareSampleRate(eCommunications));
   EXPECT_GE(fs, 0);
 
   // Multimedia device for music, movies and live music recording.
   fs = static_cast<int>(
       WASAPIAudioOutputStream::HardwareSampleRate(eMultimedia));
   EXPECT_GE(fs, 0);
 }
 
 // Test Create(), Close() calling sequence.
 TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestCreateAndClose) {
-  if (!CanRunAudioTests())
+  scoped_refptr<AudioManager> audio_manager(AudioManager::Create());
+  if (!CanRunAudioTests(audio_manager.get()))
     return;
-  AudioOutputStream* aos = CreateDefaultAudioOutputStream();
+  AudioOutputStream* aos = CreateDefaultAudioOutputStream(audio_manager.get());
   aos->Close();
 }
 
 // Test Open(), Close() calling sequence.
 TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestOpenAndClose) {
-  if (!CanRunAudioTests())
+  scoped_refptr<AudioManager> audio_manager(AudioManager::Create());
+  if (!CanRunAudioTests(audio_manager.get()))
     return;
-  AudioOutputStream* aos = CreateDefaultAudioOutputStream();
+  AudioOutputStream* aos = CreateDefaultAudioOutputStream(audio_manager.get());
   EXPECT_TRUE(aos->Open());
   aos->Close();
 }
 
 // Test Open(), Start(), Close() calling sequence.
 TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestOpenStartAndClose) {
-  if (!CanRunAudioTests())
+  scoped_refptr<AudioManager> audio_manager(AudioManager::Create());
+  if (!CanRunAudioTests(audio_manager.get()))
     return;
-  AudioOutputStream* aos = CreateDefaultAudioOutputStream();
+  AudioOutputStream* aos = CreateDefaultAudioOutputStream(audio_manager.get());
   EXPECT_TRUE(aos->Open());
   MockAudioSourceCallback source;
   EXPECT_CALL(source, OnError(aos, _))
       .Times(0);
   aos->Start(&source);
   aos->Close();
 }
 
 // Test Open(), Start(), Stop(), Close() calling sequence.
 TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestOpenStartStopAndClose) {
-  if (!CanRunAudioTests())
+  scoped_refptr<AudioManager> audio_manager(AudioManager::Create());
+  if (!CanRunAudioTests(audio_manager.get()))
     return;
-  AudioOutputStream* aos = CreateDefaultAudioOutputStream();
+  AudioOutputStream* aos = CreateDefaultAudioOutputStream(audio_manager.get());
   EXPECT_TRUE(aos->Open());
   MockAudioSourceCallback source;
   EXPECT_CALL(source, OnError(aos, _))
       .Times(0);
   aos->Start(&source);
   aos->Stop();
   aos->Close();
 }
 
 // Test SetVolume(), GetVolume()
 TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestVolume) {
-  if (!CanRunAudioTests())
+  scoped_refptr<AudioManager> audio_manager(AudioManager::Create());
+  if (!CanRunAudioTests(audio_manager.get()))
     return;
-  AudioOutputStream* aos = CreateDefaultAudioOutputStream();
+  AudioOutputStream* aos = CreateDefaultAudioOutputStream(audio_manager.get());
 
   // Initial volume should be full volume (1.0).
   double volume = 0.0;
   aos->GetVolume(&volume);
   EXPECT_EQ(1.0, volume);
 
   // Verify some valid volume settings.
   aos->SetVolume(0.0);
   aos->GetVolume(&volume);
   EXPECT_EQ(0.0, volume);
@@ skipping 13 matching lines @@
 
   aos->SetVolume(-0.5);
   aos->GetVolume(&volume);
   EXPECT_EQ(1.0, volume);
 
   aos->Close();
 }
 
 // Test some additional calling sequences.
 TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestMiscCallingSequences) {
-  if (!CanRunAudioTests())
+  scoped_refptr<AudioManager> audio_manager(AudioManager::Create());
+  if (!CanRunAudioTests(audio_manager.get()))
     return;
-  AudioOutputStream* aos = CreateDefaultAudioOutputStream();
+
+  AudioOutputStream* aos = CreateDefaultAudioOutputStream(audio_manager.get());
   WASAPIAudioOutputStream* waos = static_cast<WASAPIAudioOutputStream*>(aos);
 
   // Open(), Open() is a valid calling sequence (second call does nothing).
   EXPECT_TRUE(aos->Open());
   EXPECT_TRUE(aos->Open());
 
   MockAudioSourceCallback source;
 
   // Start(), Start() is a valid calling sequence (second call does nothing).
   aos->Start(&source);
   EXPECT_TRUE(waos->started());
   aos->Start(&source);
   EXPECT_TRUE(waos->started());
 
   // Stop(), Stop() is a valid calling sequence (second call does nothing).
   aos->Stop();
   EXPECT_FALSE(waos->started());
   aos->Stop();
   EXPECT_FALSE(waos->started());
 
   aos->Close();
 }
 
 // Use default packet size (10ms) and verify that rendering starts.
 TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestPacketSizeInMilliseconds) {
-  if (!CanRunAudioTests())
+  scoped_refptr<AudioManager> audio_manager(AudioManager::Create());
+  if (!CanRunAudioTests(audio_manager.get()))
     return;
 
   MessageLoopForUI loop;
   scoped_refptr<base::MessageLoopProxy> proxy(loop.message_loop_proxy());
 
   MockAudioSourceCallback source;
 
   // Create default WASAPI output stream which plays out in stereo using
   // the shared mixing rate. The default buffer size is 10ms.
-  AudioOutputStreamWrapper aosw;
+  AudioOutputStreamWrapper aosw(audio_manager.get());
   AudioOutputStream* aos = aosw.Create();
   EXPECT_TRUE(aos->Open());
 
   // Derive the expected size in bytes of each packet.
   uint32 bytes_per_packet = aosw.channels() * aosw.samples_per_packet() *
                            (aosw.bits_per_sample() / 8);
 
   // Set up expected minimum delay estimation.
   AudioBuffersState state(0, bytes_per_packet);
 
@@ skipping 10 matching lines @@
   loop.PostDelayedTask(FROM_HERE, new MessageLoop::QuitTask(),
                        TestTimeouts::action_timeout_ms());
   loop.Run();
   aos->Stop();
   aos->Close();
 }
 
 // Use a fixed packets size (independent of sample rate) and verify
 // that rendering starts.
 TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestPacketSizeInSamples) {
-  if (!CanRunAudioTests())
+  scoped_refptr<AudioManager> audio_manager(AudioManager::Create());
+  if (!CanRunAudioTests(audio_manager.get()))
     return;
 
   MessageLoopForUI loop;
   scoped_refptr<base::MessageLoopProxy> proxy(loop.message_loop_proxy());
 
   MockAudioSourceCallback source;
 
   // Create default WASAPI output stream which plays out in stereo using
   // the shared mixing rate. The buffer size is set to 1024 samples.
-  AudioOutputStreamWrapper aosw;
+  AudioOutputStreamWrapper aosw(audio_manager.get());
   AudioOutputStream* aos = aosw.Create(1024);
   EXPECT_TRUE(aos->Open());
 
   // Derive the expected size in bytes of each packet.
   uint32 bytes_per_packet = aosw.channels() * aosw.samples_per_packet() *
                            (aosw.bits_per_sample() / 8);
 
   // Set up expected minimum delay estimation.
   AudioBuffersState state(0, bytes_per_packet);
 
   // Wait for the first callback and verify its parameters.
   EXPECT_CALL(source, OnMoreData(aos, NotNull(), bytes_per_packet,
                                  HasValidDelay(state)))
       .WillOnce(
           DoAll(
               InvokeWithoutArgs(
                   CreateFunctor(&QuitMessageLoop, proxy.get())),
               Return(bytes_per_packet)));
 
   aos->Start(&source);
   loop.PostDelayedTask(FROM_HERE, new MessageLoop::QuitTask(),
                        TestTimeouts::action_timeout_ms());
   loop.Run();
   aos->Stop();
   aos->Close();
 }
 
 TEST(WinAudioOutputTest, WASAPIAudioOutputStreamTestMono) {
-  if (!CanRunAudioTests())
+  scoped_refptr<AudioManager> audio_manager(AudioManager::Create());
+  if (!CanRunAudioTests(audio_manager.get()))
     return;
 
   MessageLoopForUI loop;
   scoped_refptr<base::MessageLoopProxy> proxy(loop.message_loop_proxy());
 
   MockAudioSourceCallback source;
 
   // Create default WASAPI output stream which plays out in *mono* using
   // the shared mixing rate. The default buffer size is 10ms.
-  AudioOutputStreamWrapper aosw;
+  AudioOutputStreamWrapper aosw(audio_manager.get());
   AudioOutputStream* aos = aosw.Create(CHANNEL_LAYOUT_MONO);
-  EXPECT_TRUE(aos->Open());
-
+  bool opened;
+  EXPECT_TRUE(opened = aos->Open());
+  if (!opened) {
+    delete aos;
+    return;
+  }
   // Derive the expected size in bytes of each packet.
   uint32 bytes_per_packet = aosw.channels() * aosw.samples_per_packet() *
                            (aosw.bits_per_sample() / 8);
 
   // Set up expected minimum delay estimation.
   AudioBuffersState state(0, bytes_per_packet);
 
   EXPECT_CALL(source, OnMoreData(aos, NotNull(), bytes_per_packet,
                                  HasValidDelay(state)))
       .WillOnce(
@@ skipping 11 matching lines @@
 }
 
 // This test is intended for manual tests and should only be enabled
 // when it is required to store the captured data on a local file.
 // By default, GTest will print out YOU HAVE 1 DISABLED TEST.
 // To include disabled tests in test execution, just invoke the test program
 // with --gtest_also_run_disabled_tests or set the GTEST_ALSO_RUN_DISABLED_TESTS
 // environment variable to a value greater than 0.
 // The test files are approximately 20 seconds long.
 TEST(WinAudioOutputTest, DISABLED_WASAPIAudioOutputStreamReadFromFile) {
-  if (!CanRunAudioTests())
+  scoped_refptr<AudioManager> audio_manager(AudioManager::Create());
+  if (!CanRunAudioTests(audio_manager.get()))
     return;
 
-  AudioOutputStreamWrapper aosw;
+  AudioOutputStreamWrapper aosw(audio_manager.get());
   AudioOutputStream* aos = aosw.Create();
   EXPECT_TRUE(aos->Open());
 
   std::string file_name;
   if (aosw.sample_rate() == 48000) {
     file_name = kSpeechFile_16b_s_48k;
   } else if (aosw.sample_rate() == 44100) {
     file_name = kSpeechFile_16b_s_44k;
   } else if (aosw.sample_rate() == 96000) {
     // Use 48kHz file at 96kHz as well. Will sound like Donald Duck.
@@ skipping 12 matching lines @@
 
   aos->Start(&file_source);
   base::PlatformThread::Sleep(file_duration_ms);
   aos->Stop();
 
   LOG(INFO) << ">> File playout has stopped.";
   aos->Close();
 }
 
 }  // namespace media