Revert "Reland "Adding ANA config event to debug dump.""
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}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 67bc899..8cb142f 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc
@@ -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 @@
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;
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 151c420..f3faf51 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h
@@ -21,8 +21,6 @@
namespace webrtc {
-class DebugDumpWriter;
-
class ControllerManager {
public:
virtual ~ControllerManager() = default;
@@ -57,18 +55,6 @@
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.
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 e638740..a2deb7b 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
@@ -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 @@
namespace webrtc {
-using ::testing::_;
using ::testing::NiceMock;
namespace {
@@ -334,50 +332,8 @@
}
}
-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);
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto
index 93b31c3..3a9eea0 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto
@@ -2,8 +2,6 @@
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 @@
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;
}
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 9fc590c..fdedf6c 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
@@ -56,13 +56,6 @@
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 @@
#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));
}
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 f4e6004..1a53d1e 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
@@ -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 @@
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
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h b/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h
index fba9ccc..a276b81 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h
@@ -27,12 +27,6 @@
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
