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 b52acce230c4d980feb25d8aa89a6f3092b619d1..814eea996745de865adbf5aed4a65a4fe5e820d7 100644 |
--- a/webrtc/modules/audio_processing/audio_processing_unittest.cc |
+++ b/webrtc/modules/audio_processing/audio_processing_unittest.cc |
@@ -20,6 +20,7 @@ |
#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" |
@@ -58,7 +59,7 @@ namespace { |
// 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 google::protobuf::int32 kChannels[] = {1, 2}; |
+const int32_t kChannels[] = {1, 2}; |
const int kSampleRates[] = {8000, 16000, 32000, 48000}; |
#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE) |
@@ -230,7 +231,7 @@ void WriteStatsMessage(const AudioProcessing::Statistic& output, |
#endif |
void OpenFileAndWriteMessage(const std::string filename, |
- const ::google::protobuf::MessageLite& msg) { |
+ const MessageLite& msg) { |
FILE* file = fopen(filename.c_str(), "wb"); |
ASSERT_TRUE(file != NULL); |
@@ -299,8 +300,7 @@ void ClearTempFiles() { |
remove(kv.second.c_str()); |
} |
-void OpenFileAndReadMessage(std::string filename, |
- ::google::protobuf::MessageLite* msg) { |
+void OpenFileAndReadMessage(std::string filename, MessageLite* msg) { |
FILE* file = fopen(filename.c_str(), "rb"); |
ASSERT_TRUE(file != NULL); |
ReadMessageFromFile(file, msg); |