Switch to standard integer types in media/.

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/memory/aligned_memory.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/audio_unittest_util.h"
 #include "media/audio/mock_audio_source_callback.h"
@@ skipping 106 matching lines @@
     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_;
-  }
+  uint32_t size() const { return size_; }
   // Returns the memory backing the file.
-  const void* GetChunkAt(uint32 offset) {
-    return &start_[offset];
-  }
+  const void* GetChunkAt(uint32_t offset) { return &start_[offset]; }
 
  private:
   HANDLE fmap_;
   char* start_;
-  uint32 size_;
+  uint32_t 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.
 
@@ skipping 96 matching lines @@
   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());
   ABORT_AUDIO_TEST_IF_NOT(audio_man->HasAudioOutputDevices());
 
-  uint32 samples_100_ms = AudioParameters::kAudioCDSampleRate / 10;
+  uint32_t 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);
@@ skipping 10 matching lines @@
 // 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;
+  uint32_t 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());
   ABORT_AUDIO_TEST_IF_NOT(audio_man->HasAudioOutputDevices());
 
-  uint32 samples_100_ms = AudioParameters::kAudioCDSampleRate / 20;
+  uint32_t 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());
@@ skipping 14 matching lines @@
 // 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());
   ABORT_AUDIO_TEST_IF_NOT(audio_man->HasAudioOutputDevices());
 
   static const int kSampleRate = 16000;
   SineWaveAudioSource source(1, 200.0, kSampleRate);
   // Compute buffer size for 100ms of audio.
-  const uint32 kSamples100ms = (kSampleRate / 1000) * 100;
+  const uint32_t 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) {
+  for (uint32_t 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());
   ABORT_AUDIO_TEST_IF_NOT(audio_man->HasAudioOutputDevices());
 
-  uint32 samples_100_ms = AudioParameters::kAudioCDSampleRate / 10;
+  uint32_t 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);
 
@@ skipping 17 matching lines @@
 // 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());
   ABORT_AUDIO_TEST_IF_NOT(audio_man->HasAudioOutputDevices());
 
   // 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;
+  uint32_t 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);
@@ skipping 13 matching lines @@
   ::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());
   ABORT_AUDIO_TEST_IF_NOT(audio_man->HasAudioOutputDevices());
 
-  uint32 samples_100_ms = AudioParameters::kAudioCDSampleRate / 10;
+  uint32_t 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;
+  uint32_t 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(), 0, 0)).WillOnce(Invoke(ClearData));
@@ skipping 32 matching lines @@
         base::AlignedAlloc(data_size_, AudioBus::kChannelAlignment)));
     audio_bus_ = AudioBus::WrapMemory(params, data_.get());
   }
   ~SyncSocketSource() override {}
 
   // AudioSourceCallback::OnMoreData implementation:
   int OnMoreData(AudioBus* audio_bus,
                  uint32_t total_bytes_delay,
                  uint32_t frames_skipped) override {
     socket_->Send(&total_bytes_delay, sizeof(total_bytes_delay));
-    uint32 size = socket_->Receive(data_.get(), data_size_);
+    uint32_t 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:
   void OnError(AudioOutputStream* stream) override {}
 
  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;
+  uint32_t 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;
 
-  uint32 total_bytes_delay = 0;
+  uint32_t total_bytes_delay = 0;
   int times = 0;
   for (int ix = 0; ix < kTwoSecFrames; ix += ctx.frames) {
     if (ctx.socket->Receive(&total_bytes_delay, sizeof(total_bytes_delay)) == 0)
       break;
     if ((times > 0) && (total_bytes_delay < 1000)) __debugbreak();
     sine.OnMoreData(audio_bus.get(), total_bytes_delay, 0);
     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());
   ABORT_AUDIO_TEST_IF_NOT(audio_man->HasAudioOutputDevices());
 
   static const int sample_rate = AudioParameters::kAudioCDSampleRate;
-  static const uint32 kSamples20ms = sample_rate / 50;
+  static const uint32_t 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());
 
@@ skipping 16 matching lines @@
   oas->Start(&source);
 
   ::WaitForSingleObject(thread, INFINITE);
   ::CloseHandle(thread);
 
   oas->Stop();
   oas->Close();
 }
 
 }  // namespace media