Revert of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #16 id:300001 of https://codereview.webrtc.org/2747863003/ )

Reason for revert:
I will try to reland next week because it is causing some problems.

Original issue's description:
> To accommodate some downstream WebRTC users we need to loosen
> the coupling between our code and the //third_party/protobuf.
>
> This includes using typedefs to define strings instead of
> assuming std::string.
>
> After this refactoring it will be possible to link with other
> protobuf implementations than the current one.
>
> We moved the PRESUBMIT check to another CL [1]. The goal of this
> presubmit is to avoid the direct usage of google::protobuf outside
> of the webrtc/base/protobuf_utils.h header file.
>
> [1] - https://codereview.webrtc.org/2753823003/
>
> BUG=webrtc:7340
> NOTRY=True
>
> Review-Url: https://codereview.webrtc.org/2747863003
> Cr-Commit-Position: refs/heads/master@{#17466}
> Committed: 16ab93b952

TBR=kjellander@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org,michaelt@webrtc.org,peah@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7340

Review-Url: https://codereview.webrtc.org/2786363002
Cr-Commit-Position: refs/heads/master@{#17483}

webrtc/BUILD.gn

@@ -100,12 +100,6 @@ config("common_config") {
   cflags_cc = []
   defines = []
 
-  if (rtc_enable_protobuf) {
-    defines += [ "WEBRTC_ENABLE_PROTOBUF=1" ]
-  } else {
-    defines += [ "WEBRTC_ENABLE_PROTOBUF=0" ]
-  }
-
   if (rtc_restrict_logging) {
     defines += [ "WEBRTC_RESTRICT_LOGGING" ]
   }

webrtc/base/BUILD.gn

@@ -79,17 +79,6 @@ if (!rtc_build_ssl) {
   }
 }
 
-source_set("protobuf_utils") {
-  sources = [
-    "protobuf_utils.h",
-  ]
-  if (rtc_enable_protobuf) {
-    public_deps = [
-      "//third_party/protobuf:protobuf_lite",
-    ]
-  }
-}
-
 # The subset of rtc_base approved for use outside of libjingle.
 rtc_static_library("rtc_base_approved") {
   defines = []

webrtc/base/DEPS

@@ -9,7 +9,4 @@ specific_include_rules = {
   "gunit_prod.h": [
     "+gtest",
   ],
-  "protobuf_utils.h": [
-    "+third_party/protobuf",
-  ],
 }

webrtc/base/protobuf_utils.h

@@ -1,36 +0,0 @@
-/*
- *  Copyright (c) 2017 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 <string>
-
-#ifndef WEBRTC_BASE_PROTOBUF_UTILS_H_
-#define WEBRTC_BASE_PROTOBUF_UTILS_H_
-
-namespace webrtc {
-
-using ProtoString = std::string;
-
-}  // namespace webrtc
-
-#if WEBRTC_ENABLE_PROTOBUF
-
-#include "third_party/protobuf/src/google/protobuf/message_lite.h"
-#include "third_party/protobuf/src/google/protobuf/repeated_field.h"
-
-namespace webrtc {
-
-using google::protobuf::MessageLite;
-using google::protobuf::RepeatedPtrField;
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_ENABLE_PROTOBUF
-
-#endif  // WEBRTC_BASE_PROTOBUF_UTILS_H_

webrtc/logging/BUILD.gn

@@ -51,7 +51,6 @@ rtc_static_library("rtc_event_log_impl") {
   deps = [
     ":rtc_event_log_api",
     "..:webrtc_common",
-    "../base:protobuf_utils",
     "../base:rtc_base_approved",
     "../call:call_interfaces",
     "../modules/audio_coding:audio_network_adaptor",
@@ -97,7 +96,6 @@ if (rtc_enable_protobuf) {
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
     deps = [
-      "../base:protobuf_utils",
       "../base:rtc_base_approved",
     ]
   }

webrtc/logging/rtc_event_log/rtc_event_log.cc

@@ -16,7 +16,6 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/event.h"
-#include "webrtc/base/protobuf_utils.h"
 #include "webrtc/base/swap_queue.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/base/timeutils.h"
@@ -38,7 +37,7 @@
 #include "webrtc/system_wrappers/include/logging.h"
 
 #ifdef ENABLE_RTC_EVENT_LOG
-// *.pb.h files are generated at build-time by the protobuf compiler.
+// Files generated at build-time by the protobuf compiler.
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
 #else
@@ -584,7 +583,7 @@ bool RtcEventLog::ParseRtcEventLog(const std::string& file_name,
   if (!dump_file->OpenFile(file_name.c_str(), true)) {
     return false;
   }
-  ProtoString dump_buffer;
+  std::string dump_buffer;
   while ((bytes_read = dump_file->Read(tmp_buffer, sizeof(tmp_buffer))) > 0) {
     dump_buffer.append(tmp_buffer, bytes_read);
   }

webrtc/logging/rtc_event_log/rtc_event_log_helper_thread.h

@@ -13,6 +13,7 @@
 
 #include <limits>
 #include <memory>
+#include <string>
 #include <utility>
 #include <vector>
 
@@ -20,13 +21,12 @@
 #include "webrtc/base/event.h"
 #include "webrtc/base/ignore_wundef.h"
 #include "webrtc/base/platform_thread.h"
-#include "webrtc/base/protobuf_utils.h"
 #include "webrtc/base/swap_queue.h"
 #include "webrtc/logging/rtc_event_log/ringbuffer.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 
 #ifdef ENABLE_RTC_EVENT_LOG
-// *.ph.h files are generated at build-time by the protobuf compiler.
+// Files generated at build-time by the protobuf compiler.
 RTC_PUSH_IGNORING_WUNDEF()
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
@@ -112,7 +112,7 @@ class RtcEventLogHelperThread final {
   std::unique_ptr<rtclog::Event> most_recent_event_;
 
   // Temporary space for serializing profobuf data.
-  ProtoString output_string_;
+  std::string output_string_;
 
   rtc::Event wake_periodically_;
   rtc::Event wake_from_hibernation_;

webrtc/logging/rtc_event_log/rtc_event_log_parser.cc

@@ -20,7 +20,6 @@
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/base/protobuf_utils.h"
 #include "webrtc/call/call.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@@ -128,8 +127,8 @@ std::pair<uint64_t, bool> ParseVarInt(std::istream& stream) {
 
 void GetHeaderExtensions(
     std::vector<RtpExtension>* header_extensions,
-    const RepeatedPtrField<rtclog::RtpHeaderExtension>&
+    const google::protobuf::RepeatedPtrField<rtclog::RtpHeaderExtension>&
-    proto_header_extensions) {
+        proto_header_extensions) {
   header_extensions->clear();
   for (auto& p : proto_header_extensions) {
     RTC_CHECK(p.has_name());

webrtc/modules/audio_coding/BUILD.gn

@@ -74,7 +74,6 @@ rtc_static_library("rent_a_codec") {
   deps = [
            "../../api/audio_codecs:audio_codecs_api",
            "../..:webrtc_common",
-           "../../base:protobuf_utils",
            "../../base:rtc_base_approved",
            "../../system_wrappers",
            ":audio_coding_module_typedefs",
@@ -83,7 +82,6 @@ rtc_static_library("rent_a_codec") {
            ":isac_fix_c",
            ":neteq_decoder_enum",
          ] + audio_codec_deps
-
   defines = audio_codec_defines
 }
 
@@ -830,7 +828,6 @@ rtc_static_library("webrtc_opus") {
     ":audio_network_adaptor",
     "../..:webrtc_common",
     "../../api/audio_codecs:audio_codecs_api",
-    "../../base:protobuf_utils",
     "../../base:rtc_base_approved",
     "../../base:rtc_numerics",
     "../../common_audio",
@@ -924,7 +921,6 @@ rtc_static_library("audio_network_adaptor") {
 
   deps = [
     "../..:webrtc_common",
-    "../../base:protobuf_utils",
     "../../base:rtc_base_approved",
     "../../common_audio",
     "../../logging:rtc_event_log_api",
@@ -1192,12 +1188,10 @@ if (rtc_include_tests) {
       ":neteq_unittest_tools",
       ":webrtc_opus",
       "../..:webrtc_common",
-      "../../base:protobuf_utils",
       "../../base:rtc_base_approved",
       "../../system_wrappers:system_wrappers",
       "../../test:test_support",
     ]
-
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
@@ -1332,7 +1326,6 @@ if (rtc_include_tests) {
       ":neteq",
       ":neteq_unittest_tools",
       "../../api/audio_codecs:audio_codecs_api",
-      "../../base:protobuf_utils",
       "../../common_audio",
       "../../test:test_main",
       "//testing/gtest",
@@ -2089,7 +2082,6 @@ if (rtc_include_tests) {
       "../..:webrtc_common",
       "../../api/audio_codecs:audio_codecs_api",
       "../../api/audio_codecs:builtin_audio_decoder_factory",
-      "../../base:protobuf_utils",
       "../../base:rtc_base",
       "../../base:rtc_base_approved",
       "../../base:rtc_base_tests_utils",

webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc

@@ -195,7 +195,7 @@ ControllerManagerImpl::Config::Config(int min_reordering_time_ms,
 ControllerManagerImpl::Config::~Config() = default;
 
 std::unique_ptr<ControllerManager> ControllerManagerImpl::Create(
-    const ProtoString& config_string,
+    const std::string& config_string,
     size_t num_encoder_channels,
     rtc::ArrayView<const int> encoder_frame_lengths_ms,
     int min_encoder_bitrate_bps,

webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h

@@ -16,7 +16,6 @@
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/base/protobuf_utils.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/controller.h"
 
 namespace webrtc {
@@ -50,7 +49,7 @@ class ControllerManagerImpl final : public ControllerManager {
   };
 
   static std::unique_ptr<ControllerManager> Create(
-      const ProtoString& config_string,
+      const std::string& config_string,
       size_t num_encoder_channels,
       rtc::ArrayView<const int> encoder_frame_lengths_ms,
       int min_encoder_bitrate_bps,

webrtc/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc

@@ -11,7 +11,6 @@
 #include <utility>
 
 #include "webrtc/base/ignore_wundef.h"
-#include "webrtc/base/protobuf_utils.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller.h"
 #include "webrtc/system_wrappers/include/clock.h"
@@ -274,7 +273,7 @@ constexpr int kInitialFrameLengthMs = 60;
 constexpr int kMinBitrateBps = 6000;
 
 ControllerManagerStates CreateControllerManager(
-    const ProtoString& config_string) {
+    const std::string& config_string) {
   ControllerManagerStates states;
   states.simulated_clock.reset(new SimulatedClock(kClockInitialTime));
   constexpr size_t kNumEncoderChannels = 2;
@@ -346,7 +345,7 @@ TEST(ControllerManagerTest, CreateFromConfigStringAndCheckDefaultOrder) {
   AddFrameLengthControllerConfig(&config);
   AddBitrateControllerConfig(&config);
 
-  ProtoString config_string;
+  std::string config_string;
   config.SerializeToString(&config_string);
 
   auto states = CreateControllerManager(config_string);
@@ -377,7 +376,7 @@ TEST(ControllerManagerTest, CreateFromConfigStringAndCheckReordering) {
 
   AddBitrateControllerConfig(&config);
 
-  ProtoString config_string;
+  std::string config_string;
   config.SerializeToString(&config_string);
 
   auto states = CreateControllerManager(config_string);

webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc

@@ -12,7 +12,6 @@
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/ignore_wundef.h"
-#include "webrtc/base/protobuf_utils.h"
 
 #ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
 RTC_PUSH_IGNORING_WUNDEF()
@@ -35,7 +34,7 @@ using audio_network_adaptor::debug_dump::EncoderRuntimeConfig;
 
 void DumpEventToFile(const Event& event, FileWrapper* dump_file) {
   RTC_CHECK(dump_file->is_open());
-  ProtoString dump_data;
+  std::string dump_data;
   event.SerializeToString(&dump_data);
   int32_t size = event.ByteSize();
   dump_file->Write(&size, sizeof(size));

webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h

@@ -12,6 +12,7 @@
 #define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP_WRITER_H_
 
 #include <memory>
+#include <string>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/controller.h"

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc

@@ -16,7 +16,6 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/numerics/exp_filter.h"
-#include "webrtc/base/protobuf_utils.h"
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/common_types.h"
@@ -193,7 +192,7 @@ AudioEncoderOpus::AudioEncoderOpus(
       audio_network_adaptor_creator_(
           audio_network_adaptor_creator
               ? std::move(audio_network_adaptor_creator)
-              : [this](const ProtoString& config_string,
+              : [this](const std::string& config_string,
                        RtcEventLog* event_log,
                        const Clock* clock) {
                   return DefaultAudioNetworkAdaptorCreator(config_string,
@@ -549,7 +548,7 @@ void AudioEncoderOpus::ApplyAudioNetworkAdaptor() {
 
 std::unique_ptr<AudioNetworkAdaptor>
 AudioEncoderOpus::DefaultAudioNetworkAdaptorCreator(
-    const ProtoString& config_string,
+    const std::string& config_string,
     RtcEventLog* event_log,
     const Clock* clock) const {
   AudioNetworkAdaptorImpl::Config config;

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h

@@ -18,7 +18,6 @@
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/optional.h"
-#include "webrtc/base/protobuf_utils.h"
 #include "webrtc/common_audio/smoothing_filter.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
@@ -157,7 +156,7 @@ class AudioEncoderOpus final : public AudioEncoder {
 
   void ApplyAudioNetworkAdaptor();
   std::unique_ptr<AudioNetworkAdaptor> DefaultAudioNetworkAdaptorCreator(
-      const ProtoString& config_string,
+      const std::string& config_string,
       RtcEventLog* event_log,
       const Clock* clock) const;
 

webrtc/modules/audio_coding/neteq/neteq_unittest.cc

@@ -25,7 +25,6 @@
 #include "webrtc/base/ignore_wundef.h"
 #include "webrtc/base/sha1digest.h"
 #include "webrtc/base/stringencode.h"
-#include "webrtc/base/protobuf_utils.h"
 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
@@ -195,7 +194,7 @@ void ResultSink::AddResult(const NetEqNetworkStatistics& stats_raw) {
   neteq_unittest::NetEqNetworkStatistics stats;
   Convert(stats_raw, &stats);
 
-  ProtoString stats_string;
+  std::string stats_string;
   ASSERT_TRUE(stats.SerializeToString(&stats_string));
   AddMessage(output_fp_, digest_.get(), stats_string);
 #else
@@ -208,7 +207,7 @@ void ResultSink::AddResult(const RtcpStatistics& stats_raw) {
   neteq_unittest::RtcpStatistics stats;
   Convert(stats_raw, &stats);
 
-  ProtoString stats_string;
+  std::string stats_string;
   ASSERT_TRUE(stats.SerializeToString(&stats_string));
   AddMessage(output_fp_, digest_.get(), stats_string);
 #else

webrtc/modules/audio_processing/BUILD.gn

@@ -232,7 +232,6 @@ rtc_static_library("audio_processing") {
     "../..:webrtc_common",
     "../../audio/utility:audio_frame_operations",
     "../../base:gtest_prod",
-    "../../base:protobuf_utils",
     "../audio_coding:isac",
   ]
   public_deps = [
@@ -525,7 +524,6 @@ if (rtc_include_tests) {
       ":audioproc_test_utils",
       "../..:webrtc_common",
       "../../base:gtest_prod",
-      "../../base:protobuf_utils",
       "../../base:rtc_base",
       "../../base:rtc_base_approved",
       "../../common_audio:common_audio",
@@ -658,10 +656,8 @@ if (rtc_include_tests) {
     deps = [
       ":audio_processing",
       ":audioproc_test_utils",
-      "../../base:protobuf_utils",
       "//testing/gtest",
     ]
-
     if (rtc_enable_intelligibility_enhancer) {
       defines = [ "WEBRTC_INTELLIGIBILITY_ENHANCER=1" ]
     } else {
@@ -682,7 +678,6 @@ if (rtc_include_tests) {
         ":audioproc_protobuf_utils",
         ":audioproc_test_utils",
         "../..:webrtc_common",
-        "../../base:protobuf_utils",
         "../../base:rtc_base_approved",
         "../../common_audio",
         "../../system_wrappers:system_wrappers_default",
@@ -707,7 +702,6 @@ if (rtc_include_tests) {
         ":audioproc_debug_proto",
         ":audioproc_protobuf_utils",
         ":audioproc_test_utils",
-        "../../base:protobuf_utils",
         "../../base:rtc_base_approved",
         "../../common_audio:common_audio",
         "../../system_wrappers",
@@ -823,7 +817,6 @@ if (rtc_include_tests) {
       deps = [
         ":audioproc_debug_proto",
         "../..:webrtc_common",
-        "../../base:protobuf_utils",
         "../../base:rtc_base_approved",
       ]
     }

webrtc/modules/audio_processing/audio_processing_impl.cc

@@ -1879,11 +1879,11 @@ int AudioProcessingImpl::WriteInitMessage() {
   audioproc::Init* msg = debug_dump_.capture.event_msg->mutable_init();
   msg->set_sample_rate(formats_.api_format.input_stream().sample_rate_hz());
 
-  msg->set_num_input_channels(static_cast<int32_t>(
+  msg->set_num_input_channels(static_cast<google::protobuf::int32>(
       formats_.api_format.input_stream().num_channels()));
-  msg->set_num_output_channels(static_cast<int32_t>(
+  msg->set_num_output_channels(static_cast<google::protobuf::int32>(
       formats_.api_format.output_stream().num_channels()));
-  msg->set_num_reverse_channels(static_cast<int32_t>(
+  msg->set_num_reverse_channels(static_cast<google::protobuf::int32>(
       formats_.api_format.reverse_input_stream().num_channels()));
   msg->set_reverse_sample_rate(
       formats_.api_format.reverse_input_stream().sample_rate_hz());
@@ -1953,7 +1953,7 @@ int AudioProcessingImpl::WriteConfigMessage(bool forced) {
   }
   config.set_experiments_description(experiments_description);
 
-  ProtoString serialized_config = config.SerializeAsString();
+  std::string serialized_config = config.SerializeAsString();
   if (!forced &&
       debug_dump_.capture.last_serialized_config == serialized_config) {
     return kNoError;

webrtc/modules/audio_processing/audio_processing_impl.h

@@ -13,13 +13,13 @@
 
 #include <list>
 #include <memory>
+#include <string>
 #include <vector>
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/function_view.h"
 #include "webrtc/base/gtest_prod_util.h"
 #include "webrtc/base/ignore_wundef.h"
-#include "webrtc/base/protobuf_utils.h"
 #include "webrtc/base/swap_queue.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
@@ -29,7 +29,7 @@
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-// *.pb.h files are generated at build-time by the protobuf compiler.
+// Files generated at build-time by the protobuf compiler.
 RTC_PUSH_IGNORING_WUNDEF()
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
@@ -200,10 +200,10 @@ class AudioProcessingImpl : public AudioProcessing {
     ApmDebugDumpThreadState();
     ~ApmDebugDumpThreadState();
     std::unique_ptr<audioproc::Event> event_msg;  // Protobuf message.
-    ProtoString event_str;  // Memory for protobuf serialization.
+    std::string event_str;  // Memory for protobuf serialization.
 
     // Serialized string of last saved APM configuration.
-    ProtoString last_serialized_config;
+    std::string last_serialized_config;
   };
 
   struct ApmDebugDumpState {

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 @@ 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 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 @@ void WriteStatsMessage(const AudioProcessing::Statistic& output,
 #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 @@ void ClearTempFiles() {
     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);

webrtc/modules/audio_processing/test/protobuf_utils.cc

@@ -30,7 +30,7 @@ size_t ReadMessageBytesFromFile(FILE* file, std::unique_ptr<uint8_t[]>* bytes) {
 }
 
 // Returns true on success, false on error or end-of-file.
-bool ReadMessageFromFile(FILE* file, MessageLite* msg) {
+bool ReadMessageFromFile(FILE* file, ::google::protobuf::MessageLite* msg) {
   std::unique_ptr<uint8_t[]> bytes;
   size_t size = ReadMessageBytesFromFile(file, &bytes);
   if (!size)

webrtc/modules/audio_processing/test/protobuf_utils.h

@@ -14,7 +14,6 @@
 #include <memory>
 
 #include "webrtc/base/ignore_wundef.h"
-#include "webrtc/base/protobuf_utils.h"
 
 RTC_PUSH_IGNORING_WUNDEF()
 #include "webrtc/modules/audio_processing/debug.pb.h"
@@ -27,7 +26,7 @@ namespace webrtc {
 size_t ReadMessageBytesFromFile(FILE* file, std::unique_ptr<uint8_t[]>* bytes);
 
 // Returns true on success, false on error or end-of-file.
-bool ReadMessageFromFile(FILE* file, MessageLite* msg);
+bool ReadMessageFromFile(FILE* file, ::google::protobuf::MessageLite* msg);
 
 }  // namespace webrtc
 

webrtc/tools/BUILD.gn

@@ -202,7 +202,6 @@ if (rtc_enable_protobuf) {
     }
     defines = [ "ENABLE_RTC_EVENT_LOG" ]
     deps = [
-      "../base:protobuf_utils",
       "../call:call_interfaces",
       "../logging:rtc_event_log_impl",
       "../logging:rtc_event_log_parser",
@@ -239,7 +238,6 @@ if (rtc_include_tests) {
       defines = [ "ENABLE_RTC_EVENT_LOG" ]
       deps = [
         ":event_log_visualizer_utils",
-        "../base:protobuf_utils",
         "../test:field_trial",
         "//third_party/gflags",
       ]
@@ -331,7 +329,6 @@ if (rtc_include_tests) {
         "$root_build_dir/{{source_file_part}}",
       ]
       deps = [
-        "../base:protobuf_utils",
         "../logging:rtc_event_log_proto",
       ]
     }