Switch OnMoreData() to use AudioBus.

media/audio/win/audio_output_win_unittest.cc

 // 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/file_util.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_util.h"
 #include "media/audio/audio_manager.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";
 
 // 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 uint32 OnMoreData(uint8* dest,
-                            uint32 max_size,
-                            AudioBuffersState buffers_state) {
+  virtual int OnMoreData(AudioBus* audio_bus,
+                         AudioBuffersState buffers_state) {
     ++callback_count_;
-    // Touch the first byte to make sure memory is good.
-    if (max_size)
-      reinterpret_cast<char*>(dest)[0] = 1;
-    return max_size;
+    // Touch the channel memory value to make sure memory is good.
+    audio_bus->Zero();
+    return audio_bus->frames();
   }
   // AudioSourceCallback::OnError implementation:
   virtual void OnError(AudioOutputStream* stream, int code) {
     ++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 detect that the AudioStream is using
-// triple buffering correctly.
-class TestSourceTripleBuffer : public TestSourceBasic {

[DaleCurtis, 2012/08/22 23:09:21]

Misleading test to begin with since it's trying to infer information about AOS
via parameters. Test would pass on AlsaOutputDevice since it always gave a new
buffer for OnMoreData.

Not possible to test this way since AudioBus is shared for the entire AOS.

[henrika (OOO until Aug 14), 2012/08/23 14:42:53]

ALSA on Windows :-)

- public:
-  TestSourceTripleBuffer() {
-    buffer_address_[0] = NULL;
-    buffer_address_[1] = NULL;
-    buffer_address_[2] = NULL;
-  }
-  // Override of TestSourceBasic::OnMoreData.
-  virtual uint32 OnMoreData(uint8* dest,
-                            uint32 max_size,
-                            AudioBuffersState buffers_state) {
-    // Call the base, which increments the callback_count_.
-    TestSourceBasic::OnMoreData(dest, max_size, buffers_state);
-    if (callback_count() % NumberOfWaveOutBuffers() == 2) {
-      set_error(!CompareExistingIfNotNULL(2, dest));
-    } else if (callback_count() % NumberOfWaveOutBuffers() == 1) {
-      set_error(!CompareExistingIfNotNULL(1, dest));
-    } else {
-      set_error(!CompareExistingIfNotNULL(0, dest));
-    }
-    if (callback_count() > kMaxNumBuffers) {
-      set_error(buffer_address_[0] == buffer_address_[1]);
-      set_error(buffer_address_[1] == buffer_address_[2]);
-    }
-    return max_size;
-  }
-
- private:
-  bool CompareExistingIfNotNULL(uint32 index, void* address) {
-    void*& entry = buffer_address_[index];
-    if (!entry)
-      entry = address;
-    return (entry == address);
-  }
-
-  void* buffer_address_[kMaxNumBuffers];
-};
-
 // 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 uint32 OnMoreData(uint8* dest,
-                            uint32 max_size,
-                            AudioBuffersState buffers_state) {
+  virtual int OnMoreData(AudioBus* audio_bus,
+                         AudioBuffersState buffers_state) {
     // Call the base, which increments the callback_count_.
-    TestSourceBasic::OnMoreData(dest, max_size, buffers_state);
+    TestSourceBasic::OnMoreData(audio_bus, buffers_state);
     if (callback_count() > kMaxNumBuffers) {
       ::Sleep(lag_in_ms_);
     }
-    return max_size;
+    return audio_bus->frames();
   }
  private:
   int laggy_after_buffer_;
   int lag_in_ms_;
 };
 
 class MockAudioSource : public AudioOutputStream::AudioSourceCallback {
  public:
-  MOCK_METHOD3(OnMoreData, uint32(uint8* dest,
-                                  uint32 max_size,
-                                  AudioBuffersState buffers_state));
+  MOCK_METHOD2(OnMoreData, int(AudioBus* audio_bus,
+                               AudioBuffersState buffers_state));
   MOCK_METHOD2(OnError, void(AudioOutputStream* stream, int code));
+
+  static int ClearData(AudioBus* audio_bus, AudioBuffersState buffers_state) {
+    audio_bus->Zero();
+    return audio_bus->frames();
+  }
 };
 
 // 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,
@@ skipping 130 matching lines @@
   }
 
   AudioOutputStream* oas = audio_man->MakeAudioOutputStream(
       AudioParameters(AudioParameters::AUDIO_PCM_LINEAR, CHANNEL_LAYOUT_STEREO,
                       8000, 16, 1024 * 1024 * 1024));
   EXPECT_TRUE(NULL == oas);
   if (oas)
     oas->Close();
 }
 
-// Test that it uses the triple buffers correctly. Because it uses the actual
-// audio device, you might hear a short pop noise for a short time.
-TEST(WinAudioTest, PCMWaveStreamTripleBuffer) {
-  scoped_ptr<AudioManager> audio_man(AudioManager::Create());
-  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));
-  ASSERT_TRUE(NULL != oas);
-  TestSourceTripleBuffer test_triple_buffer;
-  EXPECT_TRUE(oas->Open());
-  oas->Start(&test_triple_buffer);
-  ::Sleep(300);
-  EXPECT_GT(test_triple_buffer.callback_count(), kMaxNumBuffers);
-  EXPECT_FALSE(test_triple_buffer.had_error());
-  oas->Stop();
-  ::Sleep(500);
-  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::Create());
   if (!audio_man->HasAudioOutputDevices()) {
     LOG(WARNING) << "No output device detected.";
     return;
   }
 
@@ skipping 23 matching lines @@
     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));
   ASSERT_TRUE(NULL != oas);
 
-  SineWaveAudioSource source(SineWaveAudioSource::FORMAT_16BIT_LINEAR_PCM, 1,
-                             200.0, AudioParameters::kAudioCDSampleRate);
+  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::Create());
   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));
   ASSERT_TRUE(NULL != oas);
 
-  SineWaveAudioSource source(SineWaveAudioSource::FORMAT_16BIT_LINEAR_PCM, 1,
-                             200.0, AudioParameters::kAudioCDSampleRate);
+  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::Create());
   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));
   ASSERT_TRUE(NULL != oas);
 
-  SineWaveAudioSource source(SineWaveAudioSource::FORMAT_16BIT_LINEAR_PCM, 1,
-                             200.0, AudioParameters::kAudioCDSampleRate/2);
+  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 the PushSource to play a 2 seconds file clip for about 5 seconds. We
+// 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. What you hear is a sweeping tone from 1KHz
-// to 2KHz with a bit of fade out at the end for one second. The file is two
-// of these sweeping tones back to back.
+// object roughly at the same time.
 TEST(WinAudioTest, PushSourceFile16KHz)  {
   scoped_ptr<AudioManager> audio_man(AudioManager::Create());
   if (!audio_man->HasAudioOutputDevices()) {
     LOG(WARNING) << "No output device detected.";
     return;
   }
 
-  // Open sweep02_16b_mono_16KHz.raw which has no format. It contains the
-  // raw 16 bit samples for a single channel in little-endian format. The
-  // creation sample rate is 16KHz.
-  FilePath audio_file;
-  ASSERT_TRUE(PathService::Get(base::DIR_SOURCE_ROOT, &audio_file));
-  audio_file = audio_file.Append(kAudioFile1_16b_m_16K);
-  // Map the entire file in memory.
-  ReadOnlyMappedFile file_reader(audio_file.value().c_str());
-  ASSERT_TRUE(file_reader.is_valid());
-
+  static const int kSampleRate = 16000;
+  SineWaveAudioSource source(1, 200.0, kSampleRate);
   // Compute buffer size for 100ms of audio.
-  const uint32 kSamples100ms = (16000 / 1000) * 100;
-  const uint32 kSize100ms = kSamples100ms * 2;
+  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,
-                      16000, 16, kSamples100ms));
+                      kSampleRate, 16, kSamples100ms));
   ASSERT_TRUE(NULL != oas);
 
   EXPECT_TRUE(oas->Open());
 
-  uint32 offset = 0;
-  const uint32 kMaxStartOffset = file_reader.size() - kSize100ms;
+  oas->SetVolume(1.0);
+  oas->Start(&source);

[DaleCurtis, 2012/08/22 23:09:21]

Test was broken before, it never started the AOS.

 
   // 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.
-  PushSource push_source;
   for (uint32 ix = 0; ix != 100; ++ix) {
-    push_source.Write(file_reader.GetChunkAt(offset), kSize100ms);
-    if (ix == 2) {
-      // For glitch free, start playing after some buffers are in.
-      oas->Start(&push_source);
-    }
     ::Sleep(10);
-    offset += kSize100ms;
-    if (offset > kMaxStartOffset)
-      offset = 0;
+    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::Create());
   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));
   ASSERT_TRUE(NULL != oas);
 
-  SineWaveAudioSource source(SineWaveAudioSource::FORMAT_16BIT_LINEAR_PCM, 1,
-                             200.0, AudioParameters::kAudioCDSampleRate);
+  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);
@@ skipping 24 matching lines @@
   int sample_rate = static_cast<int>(media::GetAudioHardwareSampleRate());
   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));
   ASSERT_TRUE(NULL != oas);
 
-  SineWaveAudioSource source(SineWaveAudioSource::FORMAT_16BIT_LINEAR_PCM, 1,
-                             200.0, sample_rate);
+  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);
@@ skipping 25 matching lines @@
   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(), bytes_100_ms,
+  EXPECT_CALL(source, OnMoreData(NotNull(),
                                  Field(&AudioBuffersState::pending_bytes, 0)))
-      .WillOnce(Return(bytes_100_ms));
+      .WillOnce(Invoke(MockAudioSource::ClearData));
   switch (NumberOfWaveOutBuffers()) {
     case 2:
       break;  // Calls are the same as at end of 3-buffer scheme.
     case 3:
-      EXPECT_CALL(source, OnMoreData(NotNull(), bytes_100_ms,
+      EXPECT_CALL(source, OnMoreData(NotNull(),
                                      Field(&AudioBuffersState::pending_bytes,
                                            bytes_100_ms)))
-          .WillOnce(Return(bytes_100_ms));
-      EXPECT_CALL(source, OnMoreData(NotNull(), bytes_100_ms,
+          .WillOnce(Invoke(MockAudioSource::ClearData));
+      EXPECT_CALL(source, OnMoreData(NotNull(),
                                      Field(&AudioBuffersState::pending_bytes,
                                            2 * bytes_100_ms)))
-          .WillOnce(Return(bytes_100_ms));
-      EXPECT_CALL(source, OnMoreData(NotNull(), bytes_100_ms,
+          .WillOnce(Invoke(MockAudioSource::ClearData));
+      EXPECT_CALL(source, OnMoreData(NotNull(),
                                      Field(&AudioBuffersState::pending_bytes,
                                            2 * bytes_100_ms)))
           .Times(AnyNumber())
           .WillRepeatedly(Return(0));
+      break;

[DaleCurtis, 2012/08/22 23:09:21]

Not sure how this ever passed without this break;

     default:
-      ASSERT_TRUE(false) << "Unexpected number of buffers";
+      ASSERT_TRUE(false)
+          << "Unexpected number of buffers: " << NumberOfWaveOutBuffers();
   }
-  EXPECT_CALL(source, OnMoreData(NotNull(), bytes_100_ms,
+  EXPECT_CALL(source, OnMoreData(NotNull(),
                                  Field(&AudioBuffersState::pending_bytes,
                                        bytes_100_ms)))
       .Times(AnyNumber())
       .WillRepeatedly(Return(0));
-  EXPECT_CALL(source, OnMoreData(NotNull(), bytes_100_ms,
+  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:
-  explicit SyncSocketSource(base::SyncSocket* socket)
-      : socket_(socket) {}
-
-  ~SyncSocketSource() {
+  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 uint32 OnMoreData(uint8* dest,
-                            uint32 max_size,
-                            AudioBuffersState buffers_state) {
+  virtual int OnMoreData(AudioBus* audio_bus,
+                         AudioBuffersState buffers_state) {
     socket_->Send(&buffers_state, sizeof(buffers_state));
-    uint32 got = socket_->Receive(dest, max_size);
-    return got;
+    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();
   }
   // AudioSourceCallback::OnError implementation:
   virtual void OnError(AudioOutputStream* stream, int code) {
   }
 
  private:
   base::SyncSocket* socket_;
+  int data_size_;
+  scoped_ptr_malloc<float, base::ScopedPtrAlignedFree> 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));
 
-  const int kTwoSecBytes =
-      AudioParameters::kAudioCDSampleRate * 2 * sizeof(uint16);  // NOLINT
-  uint8* buffer = new uint8[kTwoSecBytes];
-  SineWaveAudioSource sine(SineWaveAudioSource::FORMAT_16BIT_LINEAR_PCM,
-                           1, ctx.sine_freq, ctx.sample_rate);
-  sine.OnMoreData(buffer, kTwoSecBytes, AudioBuffersState());
+  // Setup AudioBus wrapping data we'll pass over the sync socket.
+  scoped_ptr_malloc<float, base::ScopedPtrAlignedFree> 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 < kTwoSecBytes; ix += ctx.packet_size_bytes) {
+  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();
-    ctx.socket->Send(&buffer[ix], ctx.packet_size_bytes);
+    sine.OnMoreData(audio_bus.get(), buffers_state);
+    ctx.socket->Send(data.get(), ctx.packet_size_bytes);
     ++times;
   }
 
-  delete buffer;
   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::Create());
   if (!audio_man->HasAudioOutputDevices()) {
     LOG(WARNING) << "No output device detected.";
     return;
   }
 
-  int sample_rate = AudioParameters::kAudioCDSampleRate;
-  const uint32 kSamples20ms = sample_rate / 50;
-  AudioOutputStream* oas = audio_man->MakeAudioOutputStream(
-      AudioParameters(AudioParameters::AUDIO_PCM_LINEAR,
-                      CHANNEL_LAYOUT_MONO, sample_rate, 16, kSamples20ms));
+  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);
   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]);
+  SyncSocketSource source(&sockets[0], params);
 
   SyncThreadContext thread_context;
-  thread_context.sample_rate = sample_rate;
+  thread_context.sample_rate = params.sample_rate();
   thread_context.sine_freq = 200.0;
-  thread_context.packet_size_bytes = kSamples20ms * 2;
+  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