Switch to standard integer types in media/.

media/audio/win/audio_low_latency_input_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/environment.h"
 #include "base/files/file_util.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/message_loop/message_loop.h"
 #include "base/path_service.h"
 #include "base/test/test_timeouts.h"
 #include "base/win/scoped_com_initializer.h"
 #include "media/audio/audio_io.h"
 #include "media/audio/audio_manager_base.h"
 #include "media/audio/audio_unittest_util.h"
@@ skipping 15 matching lines @@
   if (++*count >= limit) {
     loop->PostTask(FROM_HERE, base::MessageLoop::QuitWhenIdleClosure());
   }
 }
 
 class MockAudioInputCallback : public AudioInputStream::AudioInputCallback {
  public:
   MOCK_METHOD4(OnData,
                void(AudioInputStream* stream,
                     const AudioBus* src,
-                    uint32 hardware_delay_bytes,
+                    uint32_t hardware_delay_bytes,
                     double volume));
   MOCK_METHOD1(OnError, void(AudioInputStream* stream));
 };
 
 class FakeAudioInputCallback : public AudioInputStream::AudioInputCallback {
  public:
   FakeAudioInputCallback()
       : num_received_audio_frames_(0),
         data_event_(false, false),
         error_(false) {}
 
   bool error() const { return error_; }
   int num_received_audio_frames() const { return num_received_audio_frames_; }
 
   // Waits until OnData() is called on another thread.
   void WaitForData() {
     data_event_.Wait();
   }
 
   void OnData(AudioInputStream* stream,
               const AudioBus* src,
-              uint32 hardware_delay_bytes,
+              uint32_t hardware_delay_bytes,
               double volume) override {
     EXPECT_GE(hardware_delay_bytes, 0u);
     EXPECT_LT(hardware_delay_bytes, 0xFFFFu);  // Arbitrarily picked.
     num_received_audio_frames_ += src->frames();
     data_event_.Signal();
   }
 
   void OnError(AudioInputStream* stream) override {
     error_ = true;
   }
@@ skipping 23 matching lines @@
     file_path = file_path.AppendASCII(file_name);
     binary_file_ = base::OpenFile(file_path, "wb");
     DLOG_IF(ERROR, !binary_file_) << "Failed to open binary PCM data file.";
     VLOG(0) << ">> Output file: " << file_path.value() << " has been created.";
     VLOG(0) << "bits_per_sample_:" << bits_per_sample_;
   }
 
   ~WriteToFileAudioSink() override {
     size_t bytes_written = 0;
     while (bytes_written < bytes_to_write_) {
-      const uint8* chunk;
+      const uint8_t* chunk;
       int chunk_size;
 
       // Stop writing if no more data is available.
       if (!buffer_.GetCurrentChunk(&chunk, &chunk_size))
         break;
 
       // Write recorded data chunk to the file and prepare for next chunk.
       fwrite(chunk, 1, chunk_size, binary_file_);
       buffer_.Seek(chunk_size);
       bytes_written += chunk_size;
     }
     base::CloseFile(binary_file_);
   }
 
   // AudioInputStream::AudioInputCallback implementation.
   void OnData(AudioInputStream* stream,
               const AudioBus* src,
-              uint32 hardware_delay_bytes,
+              uint32_t hardware_delay_bytes,
               double volume) override {
     EXPECT_EQ(bits_per_sample_, 16);
     const int num_samples = src->frames() * src->channels();
-    scoped_ptr<int16> interleaved(new int16[num_samples]);
+    scoped_ptr<int16_t> interleaved(new int16_t[num_samples]);
     const int bytes_per_sample = sizeof(*interleaved);
     src->ToInterleaved(src->frames(), bytes_per_sample, interleaved.get());
 
     // Store data data in a temporary buffer to avoid making blocking
     // fwrite() calls in the audio callback. The complete buffer will be
     // written to file in the destructor.
     const int size = bytes_per_sample * num_samples;
-    if (buffer_.Append((const uint8*)interleaved.get(), size)) {
+    if (buffer_.Append((const uint8_t*)interleaved.get(), size)) {
       bytes_to_write_ += size;
     }
   }
 
   void OnError(AudioInputStream* stream) override {}
 
  private:
   int bits_per_sample_;
   media::SeekableBuffer buffer_;
   FILE* binary_file_;
@@ skipping 202 matching lines @@
 
   // Create default WASAPI input stream which records in stereo using
   // the shared mixing rate. The default buffer size is 10ms.
   AudioInputStreamWrapper aisw(audio_manager.get());
   ScopedAudioInputStream ais(aisw.Create());
   EXPECT_TRUE(ais->Open());
 
   MockAudioInputCallback sink;
 
   // Derive the expected size in bytes of each recorded packet.
-  uint32 bytes_per_packet = aisw.channels() * aisw.frames_per_buffer() *
-      (aisw.bits_per_sample() / 8);
+  uint32_t bytes_per_packet =
+      aisw.channels() * aisw.frames_per_buffer() * (aisw.bits_per_sample() / 8);
 
   // We use 10ms packets and will run the test until ten packets are received.
   // All should contain valid packets of the same size and a valid delay
   // estimate.
   EXPECT_CALL(sink, OnData(ais.get(), NotNull(), _, _))
       .Times(AtLeast(10))
       .WillRepeatedly(CheckCountAndPostQuitTask(&count, 10, &loop));
   ais->Start(&sink);
   loop.Run();
   ais->Stop();
@@ skipping 88 matching lines @@
   WriteToFileAudioSink file_sink(file_name, aisw.bits_per_sample());
   VLOG(0) << ">> Speak into the default microphone while recording.";
   ais->Start(&file_sink);
   base::PlatformThread::Sleep(TestTimeouts::action_timeout());
   ais->Stop();
   VLOG(0) << ">> Recording has stopped.";
   ais.Close();
 }
 
 }  // namespace media