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

webrtc/modules/audio_processing/audio_processing_unittest.cc

 /*
  *  Copyright (c) 2012 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 <math.h>
 #include <stdio.h>
 
 #include <algorithm>
 #include <limits>
 #include <memory>
 #include <queue>
 
 #include "webrtc/base/arraysize.h"
 #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"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_processing/audio_processing_impl.h"
 #include "webrtc/modules/audio_processing/beamformer/mock_nonlinear_beamformer.h"
 #include "webrtc/modules/audio_processing/common.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/level_controller/level_controller_constants.h"
 #include "webrtc/modules/audio_processing/test/protobuf_utils.h"
@@ skipping 18 matching lines @@
 // TODO(ekmeyerson): Switch to using StreamConfig and ProcessingConfig where
 // applicable.
 
 // TODO(bjornv): This is not feasible until the functionality has been
 // re-implemented; see comment at the bottom of this file. For now, the user has
 // to hard code the |write_ref_data| value.
 // When false, this will compare the output data with the results stored to
 // 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)
 // Android doesn't support 48kHz.
 const int kProcessSampleRates[] = {8000, 16000, 32000};
 #elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
 const int kProcessSampleRates[] = {8000, 16000, 32000, 48000};
 #endif
 
 enum StreamDirection { kForward = 0, kReverse };
@@ skipping 151 matching lines @@
 void WriteStatsMessage(const AudioProcessing::Statistic& output,
                        audioproc::Test::Statistic* msg) {
   msg->set_instant(output.instant);
   msg->set_average(output.average);
   msg->set_maximum(output.maximum);
   msg->set_minimum(output.minimum);
 }
 #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);
 
   int32_t size = msg.ByteSize();
   ASSERT_GT(size, 0);
   std::unique_ptr<uint8_t[]> array(new uint8_t[size]);
   ASSERT_TRUE(msg.SerializeToArray(array.get(), size));
 
   ASSERT_EQ(1u, fwrite(&size, sizeof(size), 1, file));
   ASSERT_EQ(static_cast<size_t>(size),
@@ skipping 48 matching lines @@
   if (temp_filenames[filename].empty())
     temp_filenames[filename] = test::TempFilename(test::OutputPath(), filename);
   return temp_filenames[filename];
 }
 
 void ClearTempFiles() {
   for (auto& kv : temp_filenames)
     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);
   fclose(file);
 }
 
 // Reads a 10 ms chunk of int16 interleaved audio from the given (assumed
 // stereo) file, converts to deinterleaved float (optionally downmixing) and
 // returns the result in |cb|. Returns false if the file ended (or on error) and
 // true otherwise.
@@ skipping 2563 matching lines @@
   // TODO(peah): Remove the testing for
   // apm->capture_nonlocked_.level_controller_enabled once the value in config_
   // is instead used to activate the level controller.
   EXPECT_FALSE(apm->capture_nonlocked_.level_controller_enabled);
   EXPECT_NEAR(kTargetLcPeakLeveldBFS,
               apm->config_.level_controller.initial_peak_level_dbfs,
               std::numeric_limits<float>::epsilon());
 }
 
 }  // namespace webrtc