Switch to standard integer types in content/renderer/.

content/renderer/media/media_stream_audio_processor_unittest.cc

 // Copyright 2013 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 <stddef.h>
+#include <stdint.h>
+
 #include <vector>
 
 #include "base/files/file_path.h"
 #include "base/files/file_util.h"
 #include "base/logging.h"
+#include "base/macros.h"
 #include "base/memory/aligned_memory.h"
 #include "base/path_service.h"
 #include "base/time/time.h"
+#include "build/build_config.h"
 #include "content/public/common/media_stream_request.h"
 #include "content/renderer/media/media_stream_audio_processor.h"
 #include "content/renderer/media/media_stream_audio_processor_options.h"
 #include "content/renderer/media/mock_media_constraint_factory.h"
 #include "media/audio/audio_parameters.h"
 #include "media/base/audio_bus.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
@@ skipping 30 matching lines @@
 const int kMaxNumberOfPlayoutDataChannels = 2;
 
 void ReadDataFromSpeechFile(char* data, int length) {
   base::FilePath file;
   CHECK(PathService::Get(base::DIR_SOURCE_ROOT, &file));
   file = file.Append(FILE_PATH_LITERAL("media"))
              .Append(FILE_PATH_LITERAL("test"))
              .Append(FILE_PATH_LITERAL("data"))
              .Append(FILE_PATH_LITERAL("speech_16b_stereo_48kHz.raw"));
   DCHECK(base::PathExists(file));
-  int64 data_file_size64 = 0;
+  int64_t data_file_size64 = 0;
   DCHECK(base::GetFileSize(file, &data_file_size64));
   EXPECT_EQ(length, base::ReadFile(file, data, length));
   DCHECK(data_file_size64 > length);
 }
 
 }  // namespace
 
 class MediaStreamAudioProcessorTest : public ::testing::Test {
  public:
   MediaStreamAudioProcessorTest()
       : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 512) {
   }
 
  protected:
   // Helper method to save duplicated code.
   void ProcessDataAndVerifyFormat(MediaStreamAudioProcessor* audio_processor,
                                   int expected_output_sample_rate,
                                   int expected_output_channels,
                                   int expected_output_buffer_size) {
     // Read the audio data from a file.
     const media::AudioParameters& params = audio_processor->InputFormat();
     const int packet_size =
         params.frames_per_buffer() * 2 * params.channels();
     const size_t length = packet_size * kNumberOfPacketsForTest;
     scoped_ptr<char[]> capture_data(new char[length]);
     ReadDataFromSpeechFile(capture_data.get(), length);
-    const int16* data_ptr = reinterpret_cast<const int16*>(capture_data.get());
+    const int16_t* data_ptr =
+        reinterpret_cast<const int16_t*>(capture_data.get());
     scoped_ptr<media::AudioBus> data_bus = media::AudioBus::Create(
         params.channels(), params.frames_per_buffer());
 
     // |data_bus_playout| is used if the number of capture channels is larger
     // that max allowed playout channels. |data_bus_playout_to_use| points to
     // the AudioBus to use, either |data_bus| or |data_bus_playout|.
     scoped_ptr<media::AudioBus> data_bus_playout;
     media::AudioBus* data_bus_playout_to_use = data_bus.get();
     if (params.channels() > kMaxNumberOfPlayoutDataChannels) {
       data_bus_playout =
@@ skipping 486 matching lines @@
   ProcessDataAndVerifyFormat(audio_processor.get(),
                              kAudioProcessingSampleRate,
                              kAudioProcessingNumberOfChannel,
                              kAudioProcessingSampleRate / 100);
   // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives
   // |audio_processor|.
   audio_processor = NULL;
 }
 
 }  // namespace content