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

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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}
This commit is contained in:
mbonadei
2017-03-31 03:08:07 -07:00
committed by Commit bot
parent 124a6fcddf
commit d00aad5eb2
24 changed files with 33 additions and 117 deletions
Download Patch File Download Diff File Expand all files Collapse all files

8
webrtc/modules/audio_coding/BUILD.gn
View File

@ -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",

2
webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc
View File

@ -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,

3
webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h
View File

@ -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,

7
webrtc/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
View File

@ -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);

3
webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
View File

@ -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));

1
webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h
View File

@ -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"

5
webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
View File

@ -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;

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

@ -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;

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

@ -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

7
webrtc/modules/audio_processing/BUILD.gn
View File

@ -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",
]
}

8
webrtc/modules/audio_processing/audio_processing_impl.cc
View File

@ -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;

8
webrtc/modules/audio_processing/audio_processing_impl.h
View File

@ -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 {

8
webrtc/modules/audio_processing/audio_processing_unittest.cc
View File

@ -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);

2
webrtc/modules/audio_processing/test/protobuf_utils.cc
View File

@ -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)

3
webrtc/modules/audio_processing/test/protobuf_utils.h
View File

@ -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