Convert some more things to size_t.

webrtc/modules/audio_device/android/audio_manager_unittest.cc

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/format_macros.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_device/android/build_info.h"
 #include "webrtc/modules/audio_device/android/audio_manager.h"
 #include "webrtc/modules/audio_device/android/ensure_initialized.h"
 
 #define PRINT(...) fprintf(stderr, __VA_ARGS__);
 
 namespace webrtc {
 
 static const char kTag[] = "  ";
@@ skipping 43 matching lines @@
         audio_manager()->IsLowLatencyPlayoutSupported() ? "Yes" : "No");
 }
 
 TEST_F(AudioManagerTest, ShowAudioParameterInfo) {
   const bool low_latency_out = audio_manager()->IsLowLatencyPlayoutSupported();
   PRINT("PLAYOUT:\n");
   PRINT("%saudio layer: %s\n", kTag,
         low_latency_out ? "Low latency OpenSL" : "Java/JNI based AudioTrack");
   PRINT("%ssample rate: %d Hz\n", kTag, playout_parameters_.sample_rate());
   PRINT("%schannels: %d\n", kTag, playout_parameters_.channels());
-  PRINT("%sframes per buffer: %d <=> %.2f ms\n", kTag,
+  PRINT("%sframes per buffer: %" PRIuS " <=> %.2f ms\n", kTag,
         playout_parameters_.frames_per_buffer(),
         playout_parameters_.GetBufferSizeInMilliseconds());
   PRINT("RECORD: \n");
   PRINT("%saudio layer: %s\n", kTag, "Java/JNI based AudioRecord");
   PRINT("%ssample rate: %d Hz\n", kTag, record_parameters_.sample_rate());
   PRINT("%schannels: %d\n", kTag, record_parameters_.channels());
-  PRINT("%sframes per buffer: %d <=> %.2f ms\n", kTag,
+  PRINT("%sframes per buffer: %" PRIuS " <=> %.2f ms\n", kTag,
         record_parameters_.frames_per_buffer(),
         record_parameters_.GetBufferSizeInMilliseconds());
 }
 
 // Add device-specific information to the test for logging purposes.
 TEST_F(AudioManagerTest, ShowDeviceInfo) {
   BuildInfo build_info;
   PRINT("%smodel: %s\n", kTag, build_info.GetDeviceModel().c_str());
   PRINT("%sbrand: %s\n", kTag, build_info.GetBrand().c_str());
   PRINT("%smanufacturer: %s\n",
         kTag, build_info.GetDeviceManufacturer().c_str());
 }
 
 // Add Android build information to the test for logging purposes.
 TEST_F(AudioManagerTest, ShowBuildInfo) {
   BuildInfo build_info;
   PRINT("%sbuild release: %s\n", kTag, build_info.GetBuildRelease().c_str());
   PRINT("%sbuild id: %s\n", kTag, build_info.GetAndroidBuildId().c_str());
   PRINT("%sbuild type: %s\n", kTag, build_info.GetBuildType().c_str());
   PRINT("%sSDK version: %s\n", kTag, build_info.GetSdkVersion().c_str());
 }
 
 // Basic test of the AudioParameters class using default construction where
 // all members are set to zero.
 TEST_F(AudioManagerTest, AudioParametersWithDefaultConstruction) {
   AudioParameters params;
   EXPECT_FALSE(params.is_valid());
   EXPECT_EQ(0, params.sample_rate());
   EXPECT_EQ(0, params.channels());
-  EXPECT_EQ(0, params.frames_per_buffer());
+  EXPECT_EQ(0U, params.frames_per_buffer());
   EXPECT_EQ(0U, params.frames_per_10ms_buffer());
-  EXPECT_EQ(0, params.GetBytesPerFrame());
-  EXPECT_EQ(0, params.GetBytesPerBuffer());
+  EXPECT_EQ(0U, params.GetBytesPerFrame());
+  EXPECT_EQ(0U, params.GetBytesPerBuffer());
   EXPECT_EQ(0U, params.GetBytesPer10msBuffer());
   EXPECT_EQ(0.0f, params.GetBufferSizeInMilliseconds());
 }
 
 // Basic test of the AudioParameters class using non default construction.
 TEST_F(AudioManagerTest, AudioParametersWithNonDefaultConstruction) {
   const int kSampleRate = 48000;
   const int kChannels = 1;
-  const int kFramesPerBuffer = 480;
+  const size_t kFramesPerBuffer = 480;
   const size_t kFramesPer10msBuffer = 480;
-  const int kBytesPerFrame = 2;
+  const size_t kBytesPerFrame = 2;
   const float kBufferSizeInMs = 10.0f;
   AudioParameters params(kSampleRate, kChannels, kFramesPerBuffer);
   EXPECT_TRUE(params.is_valid());
   EXPECT_EQ(kSampleRate, params.sample_rate());
   EXPECT_EQ(kChannels, params.channels());
   EXPECT_EQ(kFramesPerBuffer, params.frames_per_buffer());
   EXPECT_EQ(static_cast<size_t>(kSampleRate / 100),
             params.frames_per_10ms_buffer());
   EXPECT_EQ(kBytesPerFrame, params.GetBytesPerFrame());
   EXPECT_EQ(kBytesPerFrame * kFramesPerBuffer, params.GetBytesPerBuffer());
   EXPECT_EQ(kBytesPerFrame * kFramesPer10msBuffer,
             params.GetBytesPer10msBuffer());
   EXPECT_EQ(kBufferSizeInMs, params.GetBufferSizeInMilliseconds());
 }
 
 }  // namespace webrtc