Reland of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #1 id:1 of https://codereview.webrtc.org/2786363002/ )

Reason for revert:
Trying to re-land after solving some related issues.

There are no changes compared to the original CL.

Original issue's description:
> 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}
> Committed: d00aad5eb2

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
NOTRY=True

Review-Url: https://codereview.webrtc.org/2791963003
Cr-Commit-Position: refs/heads/master@{#17584}

webrtc/BUILD.gn

@@ -100,6 +100,12 @@ 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

@@ -79,6 +79,17 @@ 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

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

webrtc/base/protobuf_utils.h

@@ -0,0 +1,36 @@
+/*
+ *  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

@@ -51,6 +51,7 @@ 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",
@@ -97,6 +98,7 @@ 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

@@ -17,6 +17,7 @@
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/event.h"
 #include "webrtc/base/logging.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 +39,7 @@
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 
 #ifdef ENABLE_RTC_EVENT_LOG
-// Files generated at build-time by the protobuf compiler.
+// *.pb.h files are 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 +585,7 @@ bool RtcEventLog::ParseRtcEventLog(const std::string& file_name,
   if (!dump_file->OpenFile(file_name.c_str(), true)) {
     return false;
   }
-  std::string dump_buffer;
+  ProtoString 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

@@ -13,7 +13,6 @@
 
 #include <limits>
 #include <memory>
-#include <string>
 #include <utility>
 #include <vector>
 
@@ -21,12 +20,13 @@
 #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
-// Files generated at build-time by the protobuf compiler.
+// *.ph.h files are 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.
-  std::string output_string_;
+  ProtoString output_string_;
 
   rtc::Event wake_periodically_;
   rtc::Event wake_from_hibernation_;

webrtc/logging/rtc_event_log/rtc_event_log_parser.cc

@@ -20,6 +20,7 @@
 
 #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/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
@@ -128,7 +129,7 @@ std::pair<uint64_t, bool> ParseVarInt(std::istream& stream) {
 
 void GetHeaderExtensions(
     std::vector<RtpExtension>* header_extensions,
-    const google::protobuf::RepeatedPtrField<rtclog::RtpHeaderExtension>&
+    const RepeatedPtrField<rtclog::RtpHeaderExtension>&
     proto_header_extensions) {
   header_extensions->clear();
   for (auto& p : proto_header_extensions) {

webrtc/modules/audio_coding/BUILD.gn

@@ -72,6 +72,7 @@ rtc_static_library("builtin_audio_encoder_factory") {
   ]
   deps = [
            "../..:webrtc_common",
+           "../../base:protobuf_utils",
            "../../base:rtc_base_approved",
            ":audio_encoder_factory_interface",
          ] + audio_codec_deps
@@ -101,6 +102,7 @@ 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",
@@ -109,6 +111,7 @@ rtc_static_library("rent_a_codec") {
            ":isac_fix_c",
            ":neteq_decoder_enum",
          ] + audio_codec_deps
+
   defines = audio_codec_defines
 }
 
@@ -856,6 +859,7 @@ 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",
@@ -949,6 +953,7 @@ rtc_static_library("audio_network_adaptor") {
 
   deps = [
     "../..:webrtc_common",
+    "../../base:protobuf_utils",
     "../../base:rtc_base_approved",
     "../../common_audio",
     "../../logging:rtc_event_log_api",
@@ -1216,10 +1221,12 @@ 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" ]
@@ -1354,6 +1361,7 @@ if (rtc_include_tests) {
       ":neteq",
       ":neteq_unittest_tools",
       "../../api/audio_codecs:audio_codecs_api",
+      "../../base:protobuf_utils",
       "../../common_audio",
       "../../test:test_main",
       "//testing/gtest",
@@ -2112,6 +2120,7 @@ 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 std::string& config_string,
+    const ProtoString& 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

@@ -16,6 +16,7 @@
 #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 {
@@ -49,7 +50,7 @@ class ControllerManagerImpl final : public ControllerManager {
   };
 
   static std::unique_ptr<ControllerManager> Create(
-      const std::string& config_string,
+      const ProtoString& 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

@@ -11,6 +11,7 @@
 #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"
@@ -273,7 +274,7 @@ constexpr int kInitialFrameLengthMs = 60;
 constexpr int kMinBitrateBps = 6000;
 
 ControllerManagerStates CreateControllerManager(
-    const std::string& config_string) {
+    const ProtoString& config_string) {
   ControllerManagerStates states;
   states.simulated_clock.reset(new SimulatedClock(kClockInitialTime));
   constexpr size_t kNumEncoderChannels = 2;
@@ -345,7 +346,7 @@ TEST(ControllerManagerTest, CreateFromConfigStringAndCheckDefaultOrder) {
   AddFrameLengthControllerConfig(&config);
   AddBitrateControllerConfig(&config);
 
-  std::string config_string;
+  ProtoString config_string;
   config.SerializeToString(&config_string);
 
   auto states = CreateControllerManager(config_string);
@@ -376,7 +377,7 @@ TEST(ControllerManagerTest, CreateFromConfigStringAndCheckReordering) {
 
   AddBitrateControllerConfig(&config);
 
-  std::string config_string;
+  ProtoString config_string;
   config.SerializeToString(&config_string);
 
   auto states = CreateControllerManager(config_string);

webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc

@@ -12,6 +12,7 @@
 
 #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()
@@ -34,7 +35,7 @@ using audio_network_adaptor::debug_dump::EncoderRuntimeConfig;
 
 void DumpEventToFile(const Event& event, FileWrapper* dump_file) {
   RTC_CHECK(dump_file->is_open());
-  std::string dump_data;
+  ProtoString 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

@@ -12,7 +12,6 @@
 #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

@@ -18,6 +18,7 @@
 #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/string_to_number.h"
 #include "webrtc/base/timeutils.h"
@@ -370,7 +371,7 @@ AudioEncoderOpus::AudioEncoderOpus(
       audio_network_adaptor_creator_(
           audio_network_adaptor_creator
               ? std::move(audio_network_adaptor_creator)
-              : [this](const std::string& config_string,
+              : [this](const ProtoString& config_string,
                        RtcEventLog* event_log,
                        const Clock* clock) {
                   return DefaultAudioNetworkAdaptorCreator(config_string,
@@ -721,7 +722,7 @@ void AudioEncoderOpus::ApplyAudioNetworkAdaptor() {
 
 std::unique_ptr<AudioNetworkAdaptor>
 AudioEncoderOpus::DefaultAudioNetworkAdaptorCreator(
-    const std::string& config_string,
+    const ProtoString& config_string,
     RtcEventLog* event_log,
     const Clock* clock) const {
   AudioNetworkAdaptorImpl::Config config;

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

@@ -19,6 +19,7 @@
 #include "webrtc/api/audio_codecs/audio_format.h"
 #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/audio_encoder.h"
@@ -167,7 +168,7 @@ class AudioEncoderOpus final : public AudioEncoder {
 
   void ApplyAudioNetworkAdaptor();
   std::unique_ptr<AudioNetworkAdaptor> DefaultAudioNetworkAdaptorCreator(
-      const std::string& config_string,
+      const ProtoString& config_string,
       RtcEventLog* event_log,
       const Clock* clock) const;
 

webrtc/modules/audio_coding/neteq/neteq_unittest.cc

@@ -25,6 +25,7 @@
 #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"
@@ -194,7 +195,7 @@ void ResultSink::AddResult(const NetEqNetworkStatistics& stats_raw) {
   neteq_unittest::NetEqNetworkStatistics stats;
   Convert(stats_raw, &stats);
 
-  std::string stats_string;
+  ProtoString stats_string;
   ASSERT_TRUE(stats.SerializeToString(&stats_string));
   AddMessage(output_fp_, digest_.get(), stats_string);
 #else
@@ -207,7 +208,7 @@ void ResultSink::AddResult(const RtcpStatistics& stats_raw) {
   neteq_unittest::RtcpStatistics stats;
   Convert(stats_raw, &stats);
 
-  std::string stats_string;
+  ProtoString stats_string;
   ASSERT_TRUE(stats.SerializeToString(&stats_string));
   AddMessage(output_fp_, digest_.get(), stats_string);
 #else

webrtc/modules/audio_processing/BUILD.gn

@@ -232,6 +232,7 @@ rtc_static_library("audio_processing") {
     "../..:webrtc_common",
     "../../audio/utility:audio_frame_operations",
     "../../base:gtest_prod",
+    "../../base:protobuf_utils",
     "../audio_coding:isac",
   ]
   public_deps = [
@@ -524,6 +525,7 @@ 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",
@@ -655,8 +657,10 @@ 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 {
@@ -677,6 +681,7 @@ 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",
@@ -701,6 +706,7 @@ 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",
@@ -816,6 +822,7 @@ 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

@@ -1878,11 +1878,11 @@ 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<google::protobuf::int32>(
+  msg->set_num_input_channels(static_cast<int32_t>(
       formats_.api_format.input_stream().num_channels()));
-  msg->set_num_output_channels(static_cast<google::protobuf::int32>(
+  msg->set_num_output_channels(static_cast<int32_t>(
       formats_.api_format.output_stream().num_channels()));
-  msg->set_num_reverse_channels(static_cast<google::protobuf::int32>(
+  msg->set_num_reverse_channels(static_cast<int32_t>(
       formats_.api_format.reverse_input_stream().num_channels()));
   msg->set_reverse_sample_rate(
       formats_.api_format.reverse_input_stream().sample_rate_hz());
@@ -1952,7 +1952,7 @@ int AudioProcessingImpl::WriteConfigMessage(bool forced) {
   }
   config.set_experiments_description(experiments_description);
 
-  std::string serialized_config = config.SerializeAsString();
+  ProtoString 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
-// Files generated at build-time by the protobuf compiler.
+// *.pb.h files are 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.
-    std::string event_str;  // Memory for protobuf serialization.
+    ProtoString event_str;  // Memory for protobuf serialization.
 
     // Serialized string of last saved APM configuration.
-    std::string last_serialized_config;
+    ProtoString last_serialized_config;
   };
 
   struct ApmDebugDumpState {

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,
+void OpenFileAndReadMessage(std::string filename, MessageLite* msg) {
-                            ::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, ::google::protobuf::MessageLite* msg) {
+bool ReadMessageFromFile(FILE* file, MessageLite* msg) {
   std::unique_ptr<uint8_t[]> bytes;
   size_t size = ReadMessageBytesFromFile(file, &bytes);
   if (!size)

webrtc/modules/audio_processing/test/protobuf_utils.h

@@ -14,6 +14,7 @@
 #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"
@@ -26,7 +27,7 @@ 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, ::google::protobuf::MessageLite* msg);
+bool ReadMessageFromFile(FILE* file, MessageLite* msg);
 
 }  // namespace webrtc