root/media/audio/win/audio_output_win_unittest.cc

/* [<][>][^][v][top][bottom][index][help] */

DEFINITIONS

This source file includes following definitions.
  1. ClearData
  2. had_error_
  3. OnMoreData
  4. OnMoreIOData
  5. OnError
  6. callback_count
  7. had_error
  8. set_error
  9. lag_in_ms_
  10. OnMoreData
  11. size_
  12. is_valid
  13. size
  14. GetChunkAt
  15. TEST
  16. TEST
  17. TEST
  18. TEST
  19. TEST
  20. TEST
  21. TEST
  22. TEST
  23. TEST
  24. TEST
  25. TEST
  26. TEST
  27. OnMoreData
  28. OnMoreIOData
  29. OnError
  30. SyncSocketThread
  31. TEST

// Copyright (c) 2012 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/base_paths.h"
#include "base/memory/aligned_memory.h"
#include "base/path_service.h"
#include "base/sync_socket.h"
#include "base/win/scoped_com_initializer.h"
#include "base/win/windows_version.h"
#include "media/base/limits.h"
#include "media/audio/audio_io.h"
#include "media/audio/audio_manager.h"
#include "media/audio/mock_audio_source_callback.h"
#include "media/audio/simple_sources.h"
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"

using ::testing::_;
using ::testing::AnyNumber;
using ::testing::DoAll;
using ::testing::Field;
using ::testing::Invoke;
using ::testing::InSequence;
using ::testing::NiceMock;
using ::testing::NotNull;
using ::testing::Return;

using base::win::ScopedCOMInitializer;

namespace media {

static const wchar_t kAudioFile1_16b_m_16K[]
    = L"media\\test\\data\\sweep02_16b_mono_16KHz.raw";

static int ClearData(AudioBus* audio_bus, AudioBuffersState buffers_state) {
  audio_bus->Zero();
  return audio_bus->frames();
}

// This class allows to find out if the callbacks are occurring as
// expected and if any error has been reported.
class TestSourceBasic : public AudioOutputStream::AudioSourceCallback {
 public:
  explicit TestSourceBasic()
      : callback_count_(0),
        had_error_(0) {
  }
  // AudioSourceCallback::OnMoreData implementation:
  virtual int OnMoreData(AudioBus* audio_bus,
                         AudioBuffersState buffers_state) {
    ++callback_count_;
    // Touch the channel memory value to make sure memory is good.
    audio_bus->Zero();
    return audio_bus->frames();
  }
  virtual int OnMoreIOData(AudioBus* source,
                           AudioBus* dest,
                           AudioBuffersState buffers_state) {
    NOTREACHED();
    return 0;
  }
  // AudioSourceCallback::OnError implementation:
  virtual void OnError(AudioOutputStream* stream) {
    ++had_error_;
  }
  // Returns how many times OnMoreData() has been called.
  int callback_count() const {
    return callback_count_;
  }
  // Returns how many times the OnError callback was called.
  int had_error() const {
    return had_error_;
  }

  void set_error(bool error) {
    had_error_ += error ? 1 : 0;
  }

 private:
  int callback_count_;
  int had_error_;
};

const int kMaxNumBuffers = 3;
// Specializes TestSourceBasic to simulate a source that blocks for some time
// in the OnMoreData callback.
class TestSourceLaggy : public TestSourceBasic {
 public:
  TestSourceLaggy(int laggy_after_buffer, int lag_in_ms)
      : laggy_after_buffer_(laggy_after_buffer), lag_in_ms_(lag_in_ms) {
  }
  virtual int OnMoreData(AudioBus* audio_bus,
                         AudioBuffersState buffers_state) {
    // Call the base, which increments the callback_count_.
    TestSourceBasic::OnMoreData(audio_bus, buffers_state);
    if (callback_count() > kMaxNumBuffers) {
      ::Sleep(lag_in_ms_);
    }
    return audio_bus->frames();
  }
 private:
  int laggy_after_buffer_;
  int lag_in_ms_;
};

// Helper class to memory map an entire file. The mapping is read-only. Don't
// use for gigabyte-sized files. Attempts to write to this memory generate
// memory access violations.
class ReadOnlyMappedFile {
 public:
  explicit ReadOnlyMappedFile(const wchar_t* file_name)
      : fmap_(NULL), start_(NULL), size_(0) {
    HANDLE file = ::CreateFileW(file_name, GENERIC_READ, FILE_SHARE_READ, NULL,
                                OPEN_EXISTING, FILE_ATTRIBUTE_NORMAL, NULL);
    if (INVALID_HANDLE_VALUE == file)
      return;
    fmap_ = ::CreateFileMappingW(file, NULL, PAGE_READONLY, 0, 0, NULL);
    ::CloseHandle(file);
    if (!fmap_)
      return;
    start_ = reinterpret_cast<char*>(::MapViewOfFile(fmap_, FILE_MAP_READ,
                                                     0, 0, 0));
    if (!start_)
      return;
    MEMORY_BASIC_INFORMATION mbi = {0};
    ::VirtualQuery(start_, &mbi, sizeof(mbi));
    size_ = mbi.RegionSize;
  }
  ~ReadOnlyMappedFile() {
    if (start_) {
      ::UnmapViewOfFile(start_);
      ::CloseHandle(fmap_);
    }
  }
  // Returns true if the file was successfully mapped.
  bool is_valid() const {
    return ((start_ > 0) && (size_ > 0));
  }
  // Returns the size in bytes of the mapped memory.
  uint32 size() const {
    return size_;
  }
  // Returns the memory backing the file.
  const void* GetChunkAt(uint32 offset) {
    return &start_[offset];
  }

 private:
  HANDLE fmap_;
  char* start_;
  uint32 size_;
};

// ===========================================================================
// Validation of AudioManager::AUDIO_PCM_LINEAR
//
// NOTE:
// The tests can fail on the build bots when somebody connects to them via
// remote-desktop and the rdp client installs an audio device that fails to open
// at some point, possibly when the connection goes idle.

// Test that can it be created and closed.
TEST(WinAudioTest, PCMWaveStreamGetAndClose) {
  scoped_ptr<AudioManager> audio_man(AudioManager::CreateForTesting());
  if (!audio_man->HasAudioOutputDevices()) {
    LOG(WARNING) << "No output device detected.";
    return;
  }

  AudioOutputStream* oas = audio_man->MakeAudioOutputStream(
      AudioParameters(AudioParameters::AUDIO_PCM_LINEAR, CHANNEL_LAYOUT_STEREO,
                      8000, 16, 256),
      std::string());
  ASSERT_TRUE(NULL != oas);
  oas->Close();
}

// Test that can it be cannot be created with invalid parameters.
TEST(WinAudioTest, SanityOnMakeParams) {
  scoped_ptr<AudioManager> audio_man(AudioManager::CreateForTesting());
  if (!audio_man->HasAudioOutputDevices()) {
    LOG(WARNING) << "No output device detected.";
    return;
  }

  AudioParameters::Format fmt = AudioParameters::AUDIO_PCM_LINEAR;
  EXPECT_TRUE(NULL == audio_man->MakeAudioOutputStream(
      AudioParameters(fmt, CHANNEL_LAYOUT_UNSUPPORTED, 8000, 16, 256),
      std::string()));
  EXPECT_TRUE(NULL == audio_man->MakeAudioOutputStream(
      AudioParameters(fmt, CHANNEL_LAYOUT_MONO, 1024 * 1024, 16, 256),
      std::string()));
  EXPECT_TRUE(NULL == audio_man->MakeAudioOutputStream(
      AudioParameters(fmt, CHANNEL_LAYOUT_STEREO, 8000, 80, 256),
      std::string()));
  EXPECT_TRUE(NULL == audio_man->MakeAudioOutputStream(
      AudioParameters(fmt, CHANNEL_LAYOUT_UNSUPPORTED, 8000, 16, 256),
      std::string()));
  EXPECT_TRUE(NULL == audio_man->MakeAudioOutputStream(
      AudioParameters(fmt, CHANNEL_LAYOUT_STEREO, -8000, 16, 256),
      std::string()));
  EXPECT_TRUE(NULL == audio_man->MakeAudioOutputStream(
      AudioParameters(fmt, CHANNEL_LAYOUT_MONO, 8000, 16, -100),
      std::string()));
  EXPECT_TRUE(NULL == audio_man->MakeAudioOutputStream(
      AudioParameters(fmt, CHANNEL_LAYOUT_MONO, 8000, 16, 0),
      std::string()));
  EXPECT_TRUE(NULL == audio_man->MakeAudioOutputStream(
      AudioParameters(fmt, CHANNEL_LAYOUT_MONO, 8000, 16,
                      media::limits::kMaxSamplesPerPacket + 1),
      std::string()));
}

// Test that it can be opened and closed.
TEST(WinAudioTest, PCMWaveStreamOpenAndClose) {
  scoped_ptr<AudioManager> audio_man(AudioManager::CreateForTesting());
  if (!audio_man->HasAudioOutputDevices()) {
    LOG(WARNING) << "No output device detected.";
    return;
  }

  AudioOutputStream* oas = audio_man->MakeAudioOutputStream(
      AudioParameters(AudioParameters::AUDIO_PCM_LINEAR, CHANNEL_LAYOUT_STEREO,
                      8000, 16, 256),
      std::string());
  ASSERT_TRUE(NULL != oas);
  EXPECT_TRUE(oas->Open());
  oas->Close();
}

// Test that it has a maximum packet size.
TEST(WinAudioTest, PCMWaveStreamOpenLimit) {
  scoped_ptr<AudioManager> audio_man(AudioManager::CreateForTesting());
  if (!audio_man->HasAudioOutputDevices()) {
    LOG(WARNING) << "No output device detected.";
    return;
  }

  AudioOutputStream* oas = audio_man->MakeAudioOutputStream(
      AudioParameters(AudioParameters::AUDIO_PCM_LINEAR, CHANNEL_LAYOUT_STEREO,
                      8000, 16, 1024 * 1024 * 1024),
      std::string());
  EXPECT_TRUE(NULL == oas);
  if (oas)
    oas->Close();
}

// Test potential deadlock situation if the source is slow or blocks for some
// time. The actual EXPECT_GT are mostly meaningless and the real test is that
// the test completes in reasonable time.
TEST(WinAudioTest, PCMWaveSlowSource) {
  scoped_ptr<AudioManager> audio_man(AudioManager::CreateForTesting());
  if (!audio_man->HasAudioOutputDevices()) {
    LOG(WARNING) << "No output device detected.";
    return;
  }

  AudioOutputStream* oas = audio_man->MakeAudioOutputStream(
      AudioParameters(AudioParameters::AUDIO_PCM_LINEAR, CHANNEL_LAYOUT_MONO,
                      16000, 16, 256),
      std::string());
  ASSERT_TRUE(NULL != oas);
  TestSourceLaggy test_laggy(2, 90);
  EXPECT_TRUE(oas->Open());
  // The test parameters cause a callback every 32 ms and the source is
  // sleeping for 90 ms, so it is guaranteed that we run out of ready buffers.
  oas->Start(&test_laggy);
  ::Sleep(500);
  EXPECT_GT(test_laggy.callback_count(), 2);
  EXPECT_FALSE(test_laggy.had_error());
  oas->Stop();
  ::Sleep(500);
  oas->Close();
}

// Test another potential deadlock situation if the thread that calls Start()
// gets paused. This test is best when run over RDP with audio enabled. See
// bug 19276 for more details.
TEST(WinAudioTest, PCMWaveStreamPlaySlowLoop) {
  scoped_ptr<AudioManager> audio_man(AudioManager::CreateForTesting());
  if (!audio_man->HasAudioOutputDevices()) {
    LOG(WARNING) << "No output device detected.";
    return;
  }

  uint32 samples_100_ms = AudioParameters::kAudioCDSampleRate / 10;
  AudioOutputStream* oas = audio_man->MakeAudioOutputStream(
      AudioParameters(AudioParameters::AUDIO_PCM_LINEAR, CHANNEL_LAYOUT_MONO,
                      AudioParameters::kAudioCDSampleRate, 16, samples_100_ms),
      std::string());
  ASSERT_TRUE(NULL != oas);

  SineWaveAudioSource source(1, 200.0, AudioParameters::kAudioCDSampleRate);

  EXPECT_TRUE(oas->Open());
  oas->SetVolume(1.0);

  for (int ix = 0; ix != 5; ++ix) {
    oas->Start(&source);
    ::Sleep(10);
    oas->Stop();
  }
  oas->Close();
}


// This test produces actual audio for .5 seconds on the default wave
// device at 44.1K s/sec. Parameters have been chosen carefully so you should
// not hear pops or noises while the sound is playing.
TEST(WinAudioTest, PCMWaveStreamPlay200HzTone44Kss) {
  scoped_ptr<AudioManager> audio_man(AudioManager::CreateForTesting());
  if (!audio_man->HasAudioOutputDevices()) {
    LOG(WARNING) << "No output device detected.";
    return;
  }

  uint32 samples_100_ms = AudioParameters::kAudioCDSampleRate / 10;
  AudioOutputStream* oas = audio_man->MakeAudioOutputStream(
      AudioParameters(AudioParameters::AUDIO_PCM_LINEAR, CHANNEL_LAYOUT_MONO,
                      AudioParameters::kAudioCDSampleRate, 16, samples_100_ms),
      std::string());
  ASSERT_TRUE(NULL != oas);

  SineWaveAudioSource source(1, 200.0, AudioParameters::kAudioCDSampleRate);

  EXPECT_TRUE(oas->Open());
  oas->SetVolume(1.0);
  oas->Start(&source);
  ::Sleep(500);
  oas->Stop();
  oas->Close();
}

// This test produces actual audio for for .5 seconds on the default wave
// device at 22K s/sec. Parameters have been chosen carefully so you should
// not hear pops or noises while the sound is playing. The audio also should
// sound with a lower volume than PCMWaveStreamPlay200HzTone44Kss.
TEST(WinAudioTest, PCMWaveStreamPlay200HzTone22Kss) {
  scoped_ptr<AudioManager> audio_man(AudioManager::CreateForTesting());
  if (!audio_man->HasAudioOutputDevices()) {
    LOG(WARNING) << "No output device detected.";
    return;
  }

  uint32 samples_100_ms = AudioParameters::kAudioCDSampleRate / 20;
  AudioOutputStream* oas = audio_man->MakeAudioOutputStream(
      AudioParameters(AudioParameters::AUDIO_PCM_LINEAR, CHANNEL_LAYOUT_MONO,
                      AudioParameters::kAudioCDSampleRate / 2, 16,
                      samples_100_ms),
      std::string());
  ASSERT_TRUE(NULL != oas);

  SineWaveAudioSource source(1, 200.0, AudioParameters::kAudioCDSampleRate/2);

  EXPECT_TRUE(oas->Open());

  oas->SetVolume(0.5);
  oas->Start(&source);
  ::Sleep(500);

  // Test that the volume is within the set limits.
  double volume = 0.0;
  oas->GetVolume(&volume);
  EXPECT_LT(volume, 0.51);
  EXPECT_GT(volume, 0.49);
  oas->Stop();
  oas->Close();
}

// Uses a restricted source to play ~2 seconds of audio for about 5 seconds. We
// try hard to generate situation where the two threads are accessing the
// object roughly at the same time.
TEST(WinAudioTest, PushSourceFile16KHz)  {
  scoped_ptr<AudioManager> audio_man(AudioManager::CreateForTesting());
  if (!audio_man->HasAudioOutputDevices()) {
    LOG(WARNING) << "No output device detected.";
    return;
  }

  static const int kSampleRate = 16000;
  SineWaveAudioSource source(1, 200.0, kSampleRate);
  // Compute buffer size for 100ms of audio.
  const uint32 kSamples100ms = (kSampleRate / 1000) * 100;
  // Restrict SineWaveAudioSource to 100ms of samples.
  source.CapSamples(kSamples100ms);

  AudioOutputStream* oas = audio_man->MakeAudioOutputStream(
      AudioParameters(AudioParameters::AUDIO_PCM_LINEAR, CHANNEL_LAYOUT_MONO,
                      kSampleRate, 16, kSamples100ms),
      std::string());
  ASSERT_TRUE(NULL != oas);

  EXPECT_TRUE(oas->Open());

  oas->SetVolume(1.0);
  oas->Start(&source);

  // We buffer and play at the same time, buffering happens every ~10ms and the
  // consuming of the buffer happens every ~100ms. We do 100 buffers which
  // effectively wrap around the file more than once.
  for (uint32 ix = 0; ix != 100; ++ix) {
    ::Sleep(10);
    source.Reset();
  }

  // Play a little bit more of the file.
  ::Sleep(500);

  oas->Stop();
  oas->Close();
}

// This test is to make sure an AudioOutputStream can be started after it was
// stopped. You will here two .5 seconds wave signal separated by 0.5 seconds
// of silence.
TEST(WinAudioTest, PCMWaveStreamPlayTwice200HzTone44Kss) {
  scoped_ptr<AudioManager> audio_man(AudioManager::CreateForTesting());
  if (!audio_man->HasAudioOutputDevices()) {
    LOG(WARNING) << "No output device detected.";
    return;
  }

  uint32 samples_100_ms = AudioParameters::kAudioCDSampleRate / 10;
  AudioOutputStream* oas = audio_man->MakeAudioOutputStream(
      AudioParameters(AudioParameters::AUDIO_PCM_LINEAR, CHANNEL_LAYOUT_MONO,
                      AudioParameters::kAudioCDSampleRate, 16, samples_100_ms),
      std::string());
  ASSERT_TRUE(NULL != oas);

  SineWaveAudioSource source(1, 200.0, AudioParameters::kAudioCDSampleRate);
  EXPECT_TRUE(oas->Open());
  oas->SetVolume(1.0);

  // Play the wave for .5 seconds.
  oas->Start(&source);
  ::Sleep(500);
  oas->Stop();

  // Sleep to give silence after stopping the AudioOutputStream.
  ::Sleep(250);

  // Start again and play for .5 seconds.
  oas->Start(&source);
  ::Sleep(500);
  oas->Stop();

  oas->Close();
}

// With the low latency mode, WASAPI is utilized by default for Vista and
// higher and Wave is used for XP and lower. It is possible to utilize a
// smaller buffer size for WASAPI than for Wave.
TEST(WinAudioTest, PCMWaveStreamPlay200HzToneLowLatency) {
  scoped_ptr<AudioManager> audio_man(AudioManager::CreateForTesting());
  if (!audio_man->HasAudioOutputDevices()) {
    LOG(WARNING) << "No output device detected.";
    return;
  }

  // The WASAPI API requires a correct COM environment.
  ScopedCOMInitializer com_init(ScopedCOMInitializer::kMTA);

  // Use 10 ms buffer size for WASAPI and 50 ms buffer size for Wave.
  // Take the existing native sample rate into account.
  const AudioParameters params = audio_man->GetDefaultOutputStreamParameters();
  int sample_rate = params.sample_rate();
  uint32 samples_10_ms = sample_rate / 100;
  int n = 1;
  (base::win::GetVersion() <= base::win::VERSION_XP) ? n = 5 : n = 1;
  AudioOutputStream* oas = audio_man->MakeAudioOutputStream(
      AudioParameters(AudioParameters::AUDIO_PCM_LOW_LATENCY,
                      CHANNEL_LAYOUT_MONO, sample_rate,
                      16, n * samples_10_ms),
      std::string());
  ASSERT_TRUE(NULL != oas);

  SineWaveAudioSource source(1, 200, sample_rate);

  bool opened = oas->Open();
  if (!opened) {
    // It was not possible to open this audio device in mono.
    // No point in continuing the test so let's break here.
    LOG(WARNING) << "Mono is not supported. Skipping test.";
    oas->Close();
    return;
  }
  oas->SetVolume(1.0);

  // Play the wave for .8 seconds.
  oas->Start(&source);
  ::Sleep(800);
  oas->Stop();
  oas->Close();
}

// Check that the pending bytes value is correct what the stream starts.
TEST(WinAudioTest, PCMWaveStreamPendingBytes) {
  scoped_ptr<AudioManager> audio_man(AudioManager::CreateForTesting());
  if (!audio_man->HasAudioOutputDevices()) {
    LOG(WARNING) << "No output device detected.";
    return;
  }

  uint32 samples_100_ms = AudioParameters::kAudioCDSampleRate / 10;
  AudioOutputStream* oas = audio_man->MakeAudioOutputStream(
      AudioParameters(AudioParameters::AUDIO_PCM_LINEAR, CHANNEL_LAYOUT_MONO,
                      AudioParameters::kAudioCDSampleRate, 16, samples_100_ms),
      std::string());
  ASSERT_TRUE(NULL != oas);

  NiceMock<MockAudioSourceCallback> source;
  EXPECT_TRUE(oas->Open());

  uint32 bytes_100_ms = samples_100_ms * 2;

  // Audio output stream has either a double or triple buffer scheme.
  // We expect the amount of pending bytes will reaching up to 2 times of
  // |bytes_100_ms| depending on number of buffers used.
  // From that it would decrease as we are playing the data but not providing
  // new one. And then we will try to provide zero data so the amount of
  // pending bytes will go down and eventually read zero.
  InSequence s;

  EXPECT_CALL(source, OnMoreData(NotNull(),
                                 Field(&AudioBuffersState::pending_bytes, 0)))
      .WillOnce(Invoke(ClearData));

  // Note: If AudioManagerWin::NumberOfWaveOutBuffers() ever changes, or if this
  // test is run on Vista, these expectations will fail.
  EXPECT_CALL(source, OnMoreData(NotNull(),
                                 Field(&AudioBuffersState::pending_bytes,
                                       bytes_100_ms)))
      .WillOnce(Invoke(ClearData));
  EXPECT_CALL(source, OnMoreData(NotNull(),
                                 Field(&AudioBuffersState::pending_bytes,
                                       2 * bytes_100_ms)))
      .WillOnce(Invoke(ClearData));
  EXPECT_CALL(source, OnMoreData(NotNull(),
                                 Field(&AudioBuffersState::pending_bytes,
                                       2 * bytes_100_ms)))
      .Times(AnyNumber())
      .WillRepeatedly(Return(0));
  EXPECT_CALL(source, OnMoreData(NotNull(),
                                 Field(&AudioBuffersState::pending_bytes,
                                       bytes_100_ms)))
      .Times(AnyNumber())
      .WillRepeatedly(Return(0));
  EXPECT_CALL(source, OnMoreData(NotNull(),
                                 Field(&AudioBuffersState::pending_bytes, 0)))
      .Times(AnyNumber())
      .WillRepeatedly(Return(0));

  oas->Start(&source);
  ::Sleep(500);
  oas->Stop();
  oas->Close();
}

// Simple source that uses a SyncSocket to retrieve the audio data
// from a potentially remote thread.
class SyncSocketSource : public AudioOutputStream::AudioSourceCallback {
 public:
  SyncSocketSource(base::SyncSocket* socket, const AudioParameters& params)
      : socket_(socket) {
    // Setup AudioBus wrapping data we'll receive over the sync socket.
    data_size_ = AudioBus::CalculateMemorySize(params);
    data_.reset(static_cast<float*>(
        base::AlignedAlloc(data_size_, AudioBus::kChannelAlignment)));
    audio_bus_ = AudioBus::WrapMemory(params, data_.get());
  }
  ~SyncSocketSource() {}

  // AudioSourceCallback::OnMoreData implementation:
  virtual int OnMoreData(AudioBus* audio_bus,
                         AudioBuffersState buffers_state) {
    socket_->Send(&buffers_state, sizeof(buffers_state));
    uint32 size = socket_->Receive(data_.get(), data_size_);
    DCHECK_EQ(static_cast<size_t>(size) % sizeof(*audio_bus_->channel(0)), 0U);
    audio_bus_->CopyTo(audio_bus);
    return audio_bus_->frames();
  }
  virtual int OnMoreIOData(AudioBus* source,
                           AudioBus* dest,
                           AudioBuffersState buffers_state) {
    NOTREACHED();
    return 0;
  }
  // AudioSourceCallback::OnError implementation:
  virtual void OnError(AudioOutputStream* stream) {
  }

 private:
  base::SyncSocket* socket_;
  int data_size_;
  scoped_ptr<float, base::AlignedFreeDeleter> data_;
  scoped_ptr<AudioBus> audio_bus_;
};

struct SyncThreadContext {
  base::SyncSocket* socket;
  int sample_rate;
  int channels;
  int frames;
  double sine_freq;
  uint32 packet_size_bytes;
};

// This thread provides the data that the SyncSocketSource above needs
// using the other end of a SyncSocket. The protocol is as follows:
//
// SyncSocketSource ---send 4 bytes ------------> SyncSocketThread
//                  <--- audio packet ----------
//
DWORD __stdcall SyncSocketThread(void* context) {
  SyncThreadContext& ctx = *(reinterpret_cast<SyncThreadContext*>(context));

  // Setup AudioBus wrapping data we'll pass over the sync socket.
  scoped_ptr<float, base::AlignedFreeDeleter> data(static_cast<float*>(
      base::AlignedAlloc(ctx.packet_size_bytes, AudioBus::kChannelAlignment)));
  scoped_ptr<AudioBus> audio_bus = AudioBus::WrapMemory(
      ctx.channels, ctx.frames, data.get());

  SineWaveAudioSource sine(1, ctx.sine_freq, ctx.sample_rate);
  const int kTwoSecFrames = ctx.sample_rate * 2;

  AudioBuffersState buffers_state;
  int times = 0;
  for (int ix = 0; ix < kTwoSecFrames; ix += ctx.frames) {
    if (ctx.socket->Receive(&buffers_state, sizeof(buffers_state)) == 0)
      break;
    if ((times > 0) && (buffers_state.pending_bytes < 1000)) __debugbreak();
    sine.OnMoreData(audio_bus.get(), buffers_state);
    ctx.socket->Send(data.get(), ctx.packet_size_bytes);
    ++times;
  }

  return 0;
}

// Test the basic operation of AudioOutputStream used with a SyncSocket.
// The emphasis is to verify that it is possible to feed data to the audio
// layer using a source based on SyncSocket. In a real situation we would
// go for the low-latency version in combination with SyncSocket, but to keep
// the test more simple, AUDIO_PCM_LINEAR is utilized instead. The main
// principle of the test still remains and we avoid the additional complexity
// related to the two different audio-layers for AUDIO_PCM_LOW_LATENCY.
// In this test you should hear a continuous 200Hz tone for 2 seconds.
TEST(WinAudioTest, SyncSocketBasic) {
  scoped_ptr<AudioManager> audio_man(AudioManager::CreateForTesting());
  if (!audio_man->HasAudioOutputDevices()) {
    LOG(WARNING) << "No output device detected.";
    return;
  }

  static const int sample_rate = AudioParameters::kAudioCDSampleRate;
  static const uint32 kSamples20ms = sample_rate / 50;
  AudioParameters params(AudioParameters::AUDIO_PCM_LINEAR,
                         CHANNEL_LAYOUT_MONO, sample_rate, 16, kSamples20ms);


  AudioOutputStream* oas = audio_man->MakeAudioOutputStream(params,
      std::string());
  ASSERT_TRUE(NULL != oas);

  ASSERT_TRUE(oas->Open());

  base::SyncSocket sockets[2];
  ASSERT_TRUE(base::SyncSocket::CreatePair(&sockets[0], &sockets[1]));

  SyncSocketSource source(&sockets[0], params);

  SyncThreadContext thread_context;
  thread_context.sample_rate = params.sample_rate();
  thread_context.sine_freq = 200.0;
  thread_context.packet_size_bytes = AudioBus::CalculateMemorySize(params);
  thread_context.frames = params.frames_per_buffer();
  thread_context.channels = params.channels();
  thread_context.socket = &sockets[1];

  HANDLE thread = ::CreateThread(NULL, 0, SyncSocketThread,
                                 &thread_context, 0, NULL);

  oas->Start(&source);

  ::WaitForSingleObject(thread, INFINITE);
  ::CloseHandle(thread);

  oas->Stop();
  oas->Close();
}

}  // namespace media

/* [<][>][^][v][top][bottom][index][help] */