diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn index 1e8508ac63..8ae54103f1 100644 --- a/webrtc/BUILD.gn +++ b/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" ] } diff --git a/webrtc/base/BUILD.gn b/webrtc/base/BUILD.gn index ed75d0f42c..7e02a3a063 100644 --- a/webrtc/base/BUILD.gn +++ b/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 = [] diff --git a/webrtc/base/DEPS b/webrtc/base/DEPS index 6abcfb8291..bb76adfe31 100644 --- a/webrtc/base/DEPS +++ b/webrtc/base/DEPS @@ -9,7 +9,4 @@ specific_include_rules = { "gunit_prod.h": [ "+gtest", ], - "protobuf_utils.h": [ - "+third_party/protobuf", - ], } diff --git a/webrtc/base/protobuf_utils.h b/webrtc/base/protobuf_utils.h deleted file mode 100644 index 69f47cf4bf..0000000000 --- a/webrtc/base/protobuf_utils.h +++ /dev/null @@ -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 - -#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_ diff --git a/webrtc/logging/BUILD.gn b/webrtc/logging/BUILD.gn index 7d9fdeacc9..0cf5731bbc 100644 --- a/webrtc/logging/BUILD.gn +++ b/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", ] } diff --git a/webrtc/logging/rtc_event_log/rtc_event_log.cc b/webrtc/logging/rtc_event_log/rtc_event_log.cc index 376a4f678a..902ce42343 100644 --- a/webrtc/logging/rtc_event_log/rtc_event_log.cc +++ b/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); } diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_helper_thread.h b/webrtc/logging/rtc_event_log/rtc_event_log_helper_thread.h index 16faad3a14..420e5c529f 100644 --- a/webrtc/logging/rtc_event_log/rtc_event_log_helper_thread.h +++ b/webrtc/logging/rtc_event_log/rtc_event_log_helper_thread.h @@ -13,6 +13,7 @@ #include #include +#include #include #include @@ -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 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_; diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc b/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc index 855cb9731b..815308df11 100644 --- a/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc +++ b/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 ParseVarInt(std::istream& stream) { void GetHeaderExtensions( std::vector* header_extensions, - const RepeatedPtrField& - proto_header_extensions) { + const google::protobuf::RepeatedPtrField& + proto_header_extensions) { header_extensions->clear(); for (auto& p : proto_header_extensions) { RTC_CHECK(p.has_name()); diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn index 883a66940d..ea3a1b716b 100644 --- a/webrtc/modules/audio_coding/BUILD.gn +++ b/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", diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc b/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc index 1e6aff1746..b0d4aed63b 100644 --- a/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc +++ b/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 ControllerManagerImpl::Create( - const ProtoString& config_string, + const std::string& config_string, size_t num_encoder_channels, rtc::ArrayView encoder_frame_lengths_ms, int min_encoder_bitrate_bps, diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h b/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h index 0124cc2fcc..155b74913e 100644 --- a/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h +++ b/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h @@ -16,7 +16,6 @@ #include #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 Create( - const ProtoString& config_string, + const std::string& config_string, size_t num_encoder_channels, rtc::ArrayView encoder_frame_lengths_ms, int min_encoder_bitrate_bps, diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc b/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc index 292f17f174..ed96e1b9c6 100644 --- a/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc +++ b/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc @@ -11,7 +11,6 @@ #include #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); diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc index 0ee466ea6b..e0af3362b8 100644 --- a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc +++ b/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)); diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h index 711e13e117..da4b0317f6 100644 --- a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h +++ b/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 +#include #include "webrtc/base/constructormagic.h" #include "webrtc/modules/audio_coding/audio_network_adaptor/controller.h" diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc index ba9e36067d..103ec9b33f 100644 --- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc +++ b/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 AudioEncoderOpus::DefaultAudioNetworkAdaptorCreator( - const ProtoString& config_string, + const std::string& config_string, RtcEventLog* event_log, const Clock* clock) const { AudioNetworkAdaptorImpl::Config config; diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h index c756acf4b5..15ded47a91 100644 --- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h +++ b/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 DefaultAudioNetworkAdaptorCreator( - const ProtoString& config_string, + const std::string& config_string, RtcEventLog* event_log, const Clock* clock) const; diff --git a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc index 47073fb945..c52f2d6aa5 100644 --- a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc +++ b/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 diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn index 84545dd8b0..36f55756f8 100644 --- a/webrtc/modules/audio_processing/BUILD.gn +++ b/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", ] } diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc index 56da2820c6..1f73c5984a 100644 --- a/webrtc/modules/audio_processing/audio_processing_impl.cc +++ b/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( + msg->set_num_input_channels(static_cast( formats_.api_format.input_stream().num_channels())); - msg->set_num_output_channels(static_cast( + msg->set_num_output_channels(static_cast( formats_.api_format.output_stream().num_channels())); - msg->set_num_reverse_channels(static_cast( + msg->set_num_reverse_channels(static_cast( 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; diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h index 2b6e6f6d75..01b640fcfc 100644 --- a/webrtc/modules/audio_processing/audio_processing_impl.h +++ b/webrtc/modules/audio_processing/audio_processing_impl.h @@ -13,13 +13,13 @@ #include #include +#include #include #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 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 { diff --git a/webrtc/modules/audio_processing/audio_processing_unittest.cc b/webrtc/modules/audio_processing/audio_processing_unittest.cc index 814eea9967..b52acce230 100644 --- a/webrtc/modules/audio_processing/audio_processing_unittest.cc +++ b/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); diff --git a/webrtc/modules/audio_processing/test/protobuf_utils.cc b/webrtc/modules/audio_processing/test/protobuf_utils.cc index cb8adf92ff..c18a13e6ed 100644 --- a/webrtc/modules/audio_processing/test/protobuf_utils.cc +++ b/webrtc/modules/audio_processing/test/protobuf_utils.cc @@ -30,7 +30,7 @@ size_t ReadMessageBytesFromFile(FILE* file, std::unique_ptr* 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 bytes; size_t size = ReadMessageBytesFromFile(file, &bytes); if (!size) diff --git a/webrtc/modules/audio_processing/test/protobuf_utils.h b/webrtc/modules/audio_processing/test/protobuf_utils.h index 8941338917..e132c9405c 100644 --- a/webrtc/modules/audio_processing/test/protobuf_utils.h +++ b/webrtc/modules/audio_processing/test/protobuf_utils.h @@ -14,7 +14,6 @@ #include #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* 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 diff --git a/webrtc/tools/BUILD.gn b/webrtc/tools/BUILD.gn index 510bc69315..169a7f49de 100644 --- a/webrtc/tools/BUILD.gn +++ b/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", ] }