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Remove webrtc::ProtoString.

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Bug: None
Change-Id: If99a977532eda41eada25f57ff0ff6fe17085986
Reviewed-on: https://webrtc-review.googlesource.com/c/122581
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26726}
This commit is contained in:
Mirko Bonadei
2019-02-16 09:59:29 +01:00
committed by Commit Bot
parent eaf6a8cbc8
commit e45c688e67
11 changed files with 29 additions and 37 deletions
Download Patch File Download Diff File Expand all files Collapse all files

5
modules/audio_coding/BUILD.gn
View File

@ -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",

5
modules/audio_coding/audio_network_adaptor/controller_manager.cc
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@ -11,6 +11,7 @@
#include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
#include <cmath>
#include <string>
#include <utility>
#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<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,
@ -229,7 +230,7 @@ std::unique_ptr<ControllerManager> ControllerManagerImpl::Create(
}
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,

6
modules/audio_coding/audio_network_adaptor/controller_manager.h
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@ -13,11 +13,11 @@
#include <map>
#include <memory>
#include <string>
#include <vector>
#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<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,
@ -58,7 +58,7 @@ class ControllerManagerImpl final : public ControllerManager {
bool initial_dtx_enabled);
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,

16
modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
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@ -8,6 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include <string>
#include <utility>
#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<int> encoder_frame_lengths_ms = {20, 60};
@ -319,8 +319,8 @@ void CheckControllersOrder(const std::vector<Controller*>& 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);

5
modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
View File

@ -10,11 +10,12 @@
#include "modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
#include <string>
#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<int32_t>(event.ByteSizeLong());
dump_file->Write(&size, sizeof(size));

6
modules/audio_coding/codecs/opus/audio_encoder_opus.cc
View File

@ -12,6 +12,7 @@
#include <algorithm>
#include <iterator>
#include <string>
#include <utility>
#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<AudioNetworkAdaptor>
AudioEncoderOpusImpl::DefaultAudioNetworkAdaptorCreator(
const ProtoString& config_string,
const std::string& config_string,
RtcEventLog* event_log) const {
AudioNetworkAdaptorImpl::Config config;
config.event_log = event_log;

3
modules/audio_coding/codecs/opus/audio_encoder_opus.h
View File

@ -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<AudioNetworkAdaptor> DefaultAudioNetworkAdaptorCreator(
const ProtoString& config_string,
const std::string& config_string,
RtcEventLog* event_log) const;
void MaybeUpdateUplinkBandwidth();

5
modules/audio_coding/neteq/neteq_unittest.cc
View File

@ -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

4
modules/audio_processing/aec_dump/write_to_file_task.cc
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@ -10,7 +10,7 @@
#include "modules/audio_processing/aec_dump/write_to_file_task.h"
#include "rtc_base/protobuf_utils.h"
#include <string>
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();

6
rtc_base/protobuf_utils.h
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@ -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"

5
rtc_tools/network_tester/packet_logger.cc
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@ -9,8 +9,9 @@
*/
#include "rtc_tools/network_tester/packet_logger.h"
#include <string>
#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);