| 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);
|
|
|