From e45c688e67904360266818f9ad5b933672d5ae3c Mon Sep 17 00:00:00 2001 From: Mirko Bonadei Date: Sat, 16 Feb 2019 09:59:29 +0100 Subject: [PATCH] Remove webrtc::ProtoString. Bug: None Change-Id: If99a977532eda41eada25f57ff0ff6fe17085986 Reviewed-on: https://webrtc-review.googlesource.com/c/122581 Commit-Queue: Mirko Bonadei Reviewed-by: Karl Wiberg Reviewed-by: Minyue Li Cr-Commit-Position: refs/heads/master@{#26726} --- modules/audio_coding/BUILD.gn | 5 +---- .../audio_network_adaptor/controller_manager.cc | 5 +++-- .../audio_network_adaptor/controller_manager.h | 6 +++--- .../controller_manager_unittest.cc | 16 ++++++++-------- .../audio_network_adaptor/debug_dump_writer.cc | 5 +++-- .../codecs/opus/audio_encoder_opus.cc | 6 +++--- .../codecs/opus/audio_encoder_opus.h | 3 +-- modules/audio_coding/neteq/neteq_unittest.cc | 5 ++--- .../aec_dump/write_to_file_task.cc | 4 ++-- rtc_base/protobuf_utils.h | 6 ------ rtc_tools/network_tester/packet_logger.cc | 5 +++-- 11 files changed, 29 insertions(+), 37 deletions(-) diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn index 2bc9635658..7999cc596b 100644 --- a/modules/audio_coding/BUILD.gn +++ b/modules/audio_coding/BUILD.gn @@ -767,6 +767,7 @@ rtc_static_library("webrtc_opus") { "../../api/audio_codecs/opus:audio_encoder_opus_config", "../../common_audio", "../../rtc_base:checks", + "../../rtc_base:protobuf_utils", "../../rtc_base:rtc_base_approved", "../../rtc_base:rtc_numerics", "../../rtc_base:safe_minmax", @@ -777,7 +778,6 @@ rtc_static_library("webrtc_opus") { ] public_deps = [ ":webrtc_opus_c", - "../../rtc_base:protobuf_utils", ] defines = audio_codec_defines @@ -1341,7 +1341,6 @@ if (rtc_include_tests) { ":neteq_test_tools", "../..:webrtc_common", "../../api/audio_codecs/opus:audio_encoder_opus", - "../../rtc_base:protobuf_utils", "../../rtc_base:rtc_base_approved", "../../system_wrappers", "../../system_wrappers:field_trial", @@ -1433,7 +1432,6 @@ if (rtc_include_tests) { "../../api/audio_codecs:audio_codecs_api", "../../api/audio_codecs/opus:audio_encoder_opus", "../../common_audio", - "../../rtc_base:protobuf_utils", "../../rtc_base/system:arch", "../../test:test_main", "//testing/gtest", @@ -2040,7 +2038,6 @@ if (rtc_include_tests) { "../../logging:rtc_event_log_api", "../../modules/rtp_rtcp:rtp_rtcp_format", "../../rtc_base:checks", - "../../rtc_base:protobuf_utils", "../../rtc_base:rtc_base", "../../rtc_base:rtc_base_approved", "../../rtc_base:rtc_base_tests_utils", diff --git a/modules/audio_coding/audio_network_adaptor/controller_manager.cc b/modules/audio_coding/audio_network_adaptor/controller_manager.cc index 91c359d796..4c0e61c6ad 100644 --- a/modules/audio_coding/audio_network_adaptor/controller_manager.cc +++ b/modules/audio_coding/audio_network_adaptor/controller_manager.cc @@ -11,6 +11,7 @@ #include "modules/audio_coding/audio_network_adaptor/controller_manager.h" #include +#include #include #include "modules/audio_coding/audio_network_adaptor/bitrate_controller.h" @@ -213,7 +214,7 @@ 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, @@ -229,7 +230,7 @@ std::unique_ptr ControllerManagerImpl::Create( } 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/modules/audio_coding/audio_network_adaptor/controller_manager.h b/modules/audio_coding/audio_network_adaptor/controller_manager.h index 49800857a5..f46450df95 100644 --- a/modules/audio_coding/audio_network_adaptor/controller_manager.h +++ b/modules/audio_coding/audio_network_adaptor/controller_manager.h @@ -13,11 +13,11 @@ #include #include +#include #include #include "modules/audio_coding/audio_network_adaptor/controller.h" #include "rtc_base/constructor_magic.h" -#include "rtc_base/protobuf_utils.h" namespace webrtc { @@ -47,7 +47,7 @@ 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, @@ -58,7 +58,7 @@ class ControllerManagerImpl final : public ControllerManager { bool initial_dtx_enabled); 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/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc b/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc index 2bdea19082..ce47699117 100644 --- a/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc +++ b/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc @@ -8,6 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include #include #include "modules/audio_coding/audio_network_adaptor/controller_manager.h" @@ -15,7 +16,6 @@ #include "modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h" #include "rtc_base/fake_clock.h" #include "rtc_base/ignore_wundef.h" -#include "rtc_base/protobuf_utils.h" #include "test/gtest.h" #if WEBRTC_ENABLE_PROTOBUF @@ -266,7 +266,7 @@ constexpr int kInitialFrameLengthMs = 60; constexpr int kMinBitrateBps = 6000; ControllerManagerStates CreateControllerManager( - const ProtoString& config_string) { + const std::string& config_string) { ControllerManagerStates states; constexpr size_t kNumEncoderChannels = 2; const std::vector encoder_frame_lengths_ms = {20, 60}; @@ -319,8 +319,8 @@ void CheckControllersOrder(const std::vector& controllers, } MATCHER_P(ControllerManagerEqual, value, "") { - ProtoString value_string; - ProtoString arg_string; + std::string value_string; + std::string arg_string; EXPECT_TRUE(arg.SerializeToString(&arg_string)); EXPECT_TRUE(value.SerializeToString(&value_string)); return arg_string == value_string; @@ -339,7 +339,7 @@ TEST(ControllerManagerTest, DebugDumpLoggedWhenCreateFromConfigString) { AddFrameLengthControllerConfig(&config); AddBitrateControllerConfig(&config); - ProtoString config_string; + std::string config_string; config.SerializeToString(&config_string); constexpr size_t kNumEncoderChannels = 2; @@ -373,7 +373,7 @@ TEST(ControllerManagerTest, CreateFromConfigStringAndCheckDefaultOrder) { AddFrameLengthControllerConfig(&config); AddBitrateControllerConfig(&config); - ProtoString config_string; + std::string config_string; config.SerializeToString(&config_string); auto states = CreateControllerManager(config_string); @@ -395,7 +395,7 @@ TEST(ControllerManagerTest, CreateCharPointFreeConfigAndCheckDefaultOrder) { AddDtxControllerConfig(&config); AddBitrateControllerConfig(&config); - ProtoString config_string; + std::string config_string; config.SerializeToString(&config_string); auto states = CreateControllerManager(config_string); @@ -426,7 +426,7 @@ TEST(ControllerManagerTest, CreateFromConfigStringAndCheckReordering) { AddBitrateControllerConfig(&config); - ProtoString config_string; + std::string config_string; config.SerializeToString(&config_string); auto states = CreateControllerManager(config_string); diff --git a/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc b/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc index 956d790685..805df0a589 100644 --- a/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc +++ b/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc @@ -10,11 +10,12 @@ #include "modules/audio_coding/audio_network_adaptor/debug_dump_writer.h" +#include + #include "absl/types/optional.h" #include "rtc_base/checks.h" #include "rtc_base/ignore_wundef.h" #include "rtc_base/numerics/safe_conversions.h" -#include "rtc_base/protobuf_utils.h" #include "rtc_base/system/file_wrapper.h" #if WEBRTC_ENABLE_PROTOBUF @@ -38,7 +39,7 @@ 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 = rtc::checked_cast(event.ByteSizeLong()); dump_file->Write(&size, sizeof(size)); diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc index c733f02106..69aa8b9acb 100644 --- a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc +++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc @@ -12,6 +12,7 @@ #include #include +#include #include #include "absl/memory/memory.h" @@ -25,7 +26,6 @@ #include "rtc_base/numerics/exp_filter.h" #include "rtc_base/numerics/safe_conversions.h" #include "rtc_base/numerics/safe_minmax.h" -#include "rtc_base/protobuf_utils.h" #include "rtc_base/string_to_number.h" #include "rtc_base/time_utils.h" #include "system_wrappers/include/field_trial.h" @@ -443,7 +443,7 @@ AudioEncoderOpusImpl::AudioEncoderOpusImpl(const AudioEncoderOpusConfig& config, : AudioEncoderOpusImpl( config, payload_type, - [this](const ProtoString& config_string, RtcEventLog* event_log) { + [this](const std::string& config_string, RtcEventLog* event_log) { return DefaultAudioNetworkAdaptorCreator(config_string, event_log); }, // We choose 5sec as initial time constant due to empirical data. @@ -870,7 +870,7 @@ void AudioEncoderOpusImpl::ApplyAudioNetworkAdaptor() { std::unique_ptr AudioEncoderOpusImpl::DefaultAudioNetworkAdaptorCreator( - const ProtoString& config_string, + const std::string& config_string, RtcEventLog* event_log) const { AudioNetworkAdaptorImpl::Config config; config.event_log = event_log; diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/modules/audio_coding/codecs/opus/audio_encoder_opus.h index 4533d733df..b970ae92a0 100644 --- a/modules/audio_coding/codecs/opus/audio_encoder_opus.h +++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.h @@ -24,7 +24,6 @@ #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h" #include "modules/audio_coding/codecs/opus/opus_interface.h" #include "rtc_base/constructor_magic.h" -#include "rtc_base/protobuf_utils.h" namespace webrtc { @@ -171,7 +170,7 @@ class AudioEncoderOpusImpl final : public AudioEncoder { void ApplyAudioNetworkAdaptor(); std::unique_ptr DefaultAudioNetworkAdaptorCreator( - const ProtoString& config_string, + const std::string& config_string, RtcEventLog* event_log) const; void MaybeUpdateUplinkBandwidth(); diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc index b9b8d08d16..b2aa578133 100644 --- a/modules/audio_coding/neteq/neteq_unittest.cc +++ b/modules/audio_coding/neteq/neteq_unittest.cc @@ -33,7 +33,6 @@ #include "rtc_base/ignore_wundef.h" #include "rtc_base/message_digest.h" #include "rtc_base/numerics/safe_conversions.h" -#include "rtc_base/protobuf_utils.h" #include "rtc_base/string_encode.h" #include "rtc_base/strings/string_builder.h" #include "rtc_base/system/arch.h" @@ -208,7 +207,7 @@ 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 @@ -221,7 +220,7 @@ 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/modules/audio_processing/aec_dump/write_to_file_task.cc b/modules/audio_processing/aec_dump/write_to_file_task.cc index d11f10b23a..4839a0927c 100644 --- a/modules/audio_processing/aec_dump/write_to_file_task.cc +++ b/modules/audio_processing/aec_dump/write_to_file_task.cc @@ -10,7 +10,7 @@ #include "modules/audio_processing/aec_dump/write_to_file_task.h" -#include "rtc_base/protobuf_utils.h" +#include namespace webrtc { @@ -39,7 +39,7 @@ void WriteToFileTask::UpdateBytesLeft(size_t event_byte_size) { } bool WriteToFileTask::Run() { - ProtoString event_string; + std::string event_string; event_.SerializeToString(&event_string); const size_t event_byte_size = event_.ByteSizeLong(); diff --git a/rtc_base/protobuf_utils.h b/rtc_base/protobuf_utils.h index 8fbc06060e..786365db1a 100644 --- a/rtc_base/protobuf_utils.h +++ b/rtc_base/protobuf_utils.h @@ -13,12 +13,6 @@ #ifndef RTC_BASE_PROTOBUF_UTILS_H_ #define RTC_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" diff --git a/rtc_tools/network_tester/packet_logger.cc b/rtc_tools/network_tester/packet_logger.cc index e62f2fbb90..eef8030981 100644 --- a/rtc_tools/network_tester/packet_logger.cc +++ b/rtc_tools/network_tester/packet_logger.cc @@ -9,8 +9,9 @@ */ #include "rtc_tools/network_tester/packet_logger.h" +#include + #include "rtc_base/checks.h" -#include "rtc_base/protobuf_utils.h" namespace webrtc { @@ -31,7 +32,7 @@ void PacketLogger::LogPacket(const NetworkTesterPacket& packet) { // | Size of the next | proto | Size of the next | ... | proto | // | proto message | message | proto message | | message | // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ - ProtoString packet_data; + std::string packet_data; packet.SerializeToString(&packet_data); RTC_DCHECK_LE(packet_data.length(), 255); RTC_DCHECK_GE(packet_data.length(), 0);