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Revert "Reland "Adding ANA config event to debug dump.""

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This reverts commit 2d54784d890be462a7fbf0fcfdc633bc4791982a.

Reason for revert: upstream conflicts

Original change's description:
> Reland "Adding ANA config event to debug dump."
> 
> Originally review in https://chromium-review.googlesource.com/c/535554/
> 
> Reverted in https://chromium-review.googlesource.com/c/539737/ due to upstreaming failure.
> 
> BUG=webrtc:7854
> 
> Change-Id: Ie4ad6ecfaf0f6b556dc662512d0be8ce94f8a4a8
> Reviewed-on: https://chromium-review.googlesource.com/541436
> Commit-Queue: Minyue Li <minyue@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#18865}

TBR=minyue@webrtc.org,ossu@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:7854
Change-Id: I28841ed088664d2965454dc52196f83c9d81773e
Reviewed-on: https://chromium-review.googlesource.com/559429
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18904}
This commit is contained in:
minyue-webrtc
2017-07-05 16:04:13 +02:00
committed by Commit Bot
parent 63d146b743
commit b16a01f14f
7 changed files with 1 additions and 129 deletions
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23
webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc
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@ -17,7 +17,6 @@
#include "webrtc/base/timeutils.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h"
@ -202,29 +201,9 @@ std::unique_ptr<ControllerManager> ControllerManagerImpl::Create(
int initial_bitrate_bps,
bool initial_fec_enabled,
bool initial_dtx_enabled) {
return Create(config_string, num_encoder_channels, encoder_frame_lengths_ms,
min_encoder_bitrate_bps, intial_channels_to_encode,
initial_frame_length_ms, initial_bitrate_bps,
initial_fec_enabled, initial_dtx_enabled, nullptr);
}
std::unique_ptr<ControllerManager> ControllerManagerImpl::Create(
const ProtoString& config_string,
size_t num_encoder_channels,
rtc::ArrayView<const int> encoder_frame_lengths_ms,
int min_encoder_bitrate_bps,
size_t intial_channels_to_encode,
int initial_frame_length_ms,
int initial_bitrate_bps,
bool initial_fec_enabled,
bool initial_dtx_enabled,
DebugDumpWriter* debug_dump_writer) {
#if WEBRTC_ENABLE_PROTOBUF
audio_network_adaptor::config::ControllerManager controller_manager_config;
RTC_CHECK(controller_manager_config.ParseFromString(config_string));
if (debug_dump_writer)
debug_dump_writer->DumpControllerManagerConfig(controller_manager_config,
rtc::TimeMillis());
controller_manager_config.ParseFromString(config_string);
std::vector<std::unique_ptr<Controller>> controllers;
std::map<const Controller*, std::pair<int, float>> scoring_points;

14
webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h
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@ -21,8 +21,6 @@
namespace webrtc {
class DebugDumpWriter;
class ControllerManager {
public:
virtual ~ControllerManager() = default;
@ -57,18 +55,6 @@ class ControllerManagerImpl final : public ControllerManager {
bool initial_fec_enabled,
bool initial_dtx_enabled);
static std::unique_ptr<ControllerManager> Create(
const ProtoString& config_string,
size_t num_encoder_channels,
rtc::ArrayView<const int> encoder_frame_lengths_ms,
int min_encoder_bitrate_bps,
size_t intial_channels_to_encode,
int initial_frame_length_ms,
int initial_bitrate_bps,
bool initial_fec_enabled,
bool initial_dtx_enabled,
DebugDumpWriter* debug_dump_writer);
explicit ControllerManagerImpl(const Config& config);
// Dependency injection for testing.

44
webrtc/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
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@ -15,7 +15,6 @@
#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/modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h"
#include "webrtc/test/gtest.h"
#if WEBRTC_ENABLE_PROTOBUF
@ -30,7 +29,6 @@ RTC_POP_IGNORING_WUNDEF()
namespace webrtc {
using ::testing::_;
using ::testing::NiceMock;
namespace {
@ -334,50 +332,8 @@ void CheckControllersOrder(const std::vector<Controller*>& controllers,
}
}
MATCHER_P(ControllerManagerEqual, value, "") {
ProtoString value_string;
ProtoString arg_string;
EXPECT_TRUE(arg.SerializeToString(&arg_string));
EXPECT_TRUE(value.SerializeToString(&value_string));
return arg_string == value_string;
}
} // namespace
TEST(ControllerManagerTest, DebugDumpLoggedWhenCreateFromConfigString) {
audio_network_adaptor::config::ControllerManager config;
config.set_min_reordering_time_ms(kMinReorderingTimeMs);
config.set_min_reordering_squared_distance(kMinReorderingSquareDistance);
AddFecControllerConfig(&config);
AddChannelControllerConfig(&config);
AddDtxControllerConfig(&config);
AddFrameLengthControllerConfig(&config);
AddBitrateControllerConfig(&config);
ProtoString config_string;
config.SerializeToString(&config_string);
constexpr size_t kNumEncoderChannels = 2;
const std::vector<int> encoder_frame_lengths_ms = {20, 60};
constexpr int64_t kClockInitialTimeMs = 12345678;
rtc::ScopedFakeClock fake_clock;
fake_clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(kClockInitialTimeMs));
auto debug_dump_writer =
std::unique_ptr<MockDebugDumpWriter>(new NiceMock<MockDebugDumpWriter>());
EXPECT_CALL(*debug_dump_writer, Die());
EXPECT_CALL(*debug_dump_writer,
DumpControllerManagerConfig(ControllerManagerEqual(config),
kClockInitialTimeMs));
ControllerManagerImpl::Create(config_string, kNumEncoderChannels,
encoder_frame_lengths_ms, kMinBitrateBps,
kIntialChannelsToEncode, kInitialFrameLengthMs,
kInitialBitrateBps, kInitialFecEnabled,
kInitialDtxEnabled, debug_dump_writer.get());
}
TEST(ControllerManagerTest, CreateFromConfigStringAndCheckDefaultOrder) {
audio_network_adaptor::config::ControllerManager config;
config.set_min_reordering_time_ms(kMinReorderingTimeMs);

5
webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto
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@ -2,8 +2,6 @@ syntax = "proto2";
option optimize_for = LITE_RUNTIME;
package webrtc.audio_network_adaptor.debug_dump;
import "config.proto";
message NetworkMetrics {
optional int32 uplink_bandwidth_bps = 1;
optional float uplink_packet_loss_fraction = 2;
@ -30,13 +28,10 @@ message Event {
enum Type {
NETWORK_METRICS = 0;
ENCODER_RUNTIME_CONFIG = 1;
CONTROLLER_MANAGER_CONFIG = 2;
}
required Type type = 1;
required uint32 timestamp = 2;
optional NetworkMetrics network_metrics = 3;
optional EncoderRuntimeConfig encoder_runtime_config = 4;
optional webrtc.audio_network_adaptor.config.ControllerManager
controller_manager_config = 5;
}

21
webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
View File

@ -56,13 +56,6 @@ class DebugDumpWriterImpl final : public DebugDumpWriter {
void DumpNetworkMetrics(const Controller::NetworkMetrics& metrics,
int64_t timestamp) override;
#if WEBRTC_ENABLE_PROTOBUF
void DumpControllerManagerConfig(
const audio_network_adaptor::config::ControllerManager&
controller_manager_config,
int64_t timestamp) override;
#endif
private:
std::unique_ptr<FileWrapper> dump_file_;
};
@ -144,20 +137,6 @@ void DebugDumpWriterImpl::DumpEncoderRuntimeConfig(
#endif // WEBRTC_ENABLE_PROTOBUF
}
#if WEBRTC_ENABLE_PROTOBUF
void DebugDumpWriterImpl::DumpControllerManagerConfig(
const audio_network_adaptor::config::ControllerManager&
controller_manager_config,
int64_t timestamp) {
Event event;
event.set_timestamp(timestamp);
event.set_type(Event::CONTROLLER_MANAGER_CONFIG);
event.mutable_controller_manager_config()->CopyFrom(
controller_manager_config);
DumpEventToFile(event, dump_file_.get());
}
#endif // WEBRTC_ENABLE_PROTOBUF
std::unique_ptr<DebugDumpWriter> DebugDumpWriter::Create(FILE* file_handle) {
return std::unique_ptr<DebugDumpWriter>(new DebugDumpWriterImpl(file_handle));
}

17
webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h
View File

@ -14,19 +14,9 @@
#include <memory>
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/ignore_wundef.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/controller.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
#include "webrtc/system_wrappers/include/file_wrapper.h"
#if WEBRTC_ENABLE_PROTOBUF
RTC_PUSH_IGNORING_WUNDEF()
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
#include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h"
#else
#include "webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h"
#endif
RTC_POP_IGNORING_WUNDEF()
#endif
namespace webrtc {
@ -41,13 +31,6 @@ class DebugDumpWriter {
virtual void DumpNetworkMetrics(const Controller::NetworkMetrics& metrics,
int64_t timestamp) = 0;
#if WEBRTC_ENABLE_PROTOBUF
virtual void DumpControllerManagerConfig(
const audio_network_adaptor::config::ControllerManager&
controller_manager_config,
int64_t timestamp) = 0;
#endif
};
} // namespace webrtc

6
webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h
View File

@ -27,12 +27,6 @@ class MockDebugDumpWriter : public DebugDumpWriter {
MOCK_METHOD2(DumpNetworkMetrics,
void(const Controller::NetworkMetrics& metrics,
int64_t timestamp));
#if WEBRTC_ENABLE_PROTOBUF
MOCK_METHOD2(DumpControllerManagerConfig,
void(const audio_network_adaptor::config::ControllerManager&
controller_manager_config,
int64_t timestamp));
#endif
};
} // namespace webrtc