Revert of Loosening the coupling between WebRTC and //third_party/protobuf

webrtc/modules/audio_processing/audio_processing_unittest.cc

Index: webrtc/modules/audio_processing/audio_processing_unittest.cc
diff --git a/webrtc/modules/audio_processing/audio_processing_unittest.cc b/webrtc/modules/audio_processing/audio_processing_unittest.cc
index 814eea996745de865adbf5aed4a65a4fe5e820d7..b52acce230c4d980feb25d8aa89a6f3092b619d1 100644
--- a/webrtc/modules/audio_processing/audio_processing_unittest.cc
+++ b/webrtc/modules/audio_processing/audio_processing_unittest.cc
@@ -20,7 +20,6 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/gtest_prod_util.h"
 #include "webrtc/base/ignore_wundef.h"
-#include "webrtc/base/protobuf_utils.h"
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_audio/resampler/push_sinc_resampler.h"
@@ -59,7 +58,7 @@
 // file. This is the typical case. When the file should be updated, it can
 // be set to true with the command-line switch --write_ref_data.
 bool write_ref_data = false;
-const int32_t kChannels[] = {1, 2};
+const google::protobuf::int32 kChannels[] = {1, 2};
 const int kSampleRates[] = {8000, 16000, 32000, 48000};
 
 #if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
@@ -231,7 +230,7 @@
 #endif
 
 void OpenFileAndWriteMessage(const std::string filename,
-                             const MessageLite& msg) {
+                             const ::google::protobuf::MessageLite& msg) {
   FILE* file = fopen(filename.c_str(), "wb");
   ASSERT_TRUE(file != NULL);
 
@@ -300,7 +299,8 @@
     remove(kv.second.c_str());
 }
 
-void OpenFileAndReadMessage(std::string filename, MessageLite* msg) {
+void OpenFileAndReadMessage(std::string filename,
+                            ::google::protobuf::MessageLite* msg) {
   FILE* file = fopen(filename.c_str(), "rb");
   ASSERT_TRUE(file != NULL);
   ReadMessageFromFile(file, msg);