Revert of Adding debug dump to audio network adaptor. (patchset #5 id:140001 of https://codereview.webrtc.org/2356763002/ )
Reason for revert:
Chromium bot fails
Original issue's description:
> Adding debug dump to audio network adaptor.
>
> BUG=webrtc:6303
>
> Committed: https://crrev.com/7e4f8928062afc8d571bb69f3223711701cbaad6
> Cr-Commit-Position: refs/heads/master@{#14361}
TBR=michaelt@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6303
Review-Url: https://codereview.webrtc.org/2362003002
Cr-Commit-Position: refs/heads/master@{#14362}
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index 5f0f47b..eaf2e74 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -709,8 +709,6 @@
"audio_network_adaptor/controller.h",
"audio_network_adaptor/controller_manager.cc",
"audio_network_adaptor/controller_manager.h",
- "audio_network_adaptor/debug_dump_writer.cc",
- "audio_network_adaptor/debug_dump_writer.h",
"audio_network_adaptor/dtx_controller.cc",
"audio_network_adaptor/dtx_controller.h",
"audio_network_adaptor/fec_controller.cc",
@@ -723,13 +721,6 @@
]
configs += [ "../..:common_config" ]
public_configs = [ "../..:common_inherited_config" ]
-
- if (rtc_enable_protobuf) {
- deps = [
- ":ana_debug_dump_proto",
- ]
- defines = [ "WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP" ]
- }
}
config("neteq_config") {
@@ -986,13 +977,6 @@
} # audio_decoder_unittests
if (rtc_enable_protobuf) {
- proto_library("ana_debug_dump_proto") {
- sources = [
- "audio_network_adaptor/debug_dump.proto",
- ]
- proto_out_dir = "webrtc/modules/audio_coding/audio_network_adaptor"
- }
-
proto_library("neteq_unittest_proto") {
sources = [
"neteq/neteq_unittest.proto",
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor.gypi b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor.gypi
index 508b62b..af6ba55 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor.gypi
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor.gypi
@@ -22,8 +22,6 @@
'controller.cc',
'controller_manager.cc',
'controller_manager.h',
- 'debug_dump_writer.cc',
- 'debug_dump_writer.h',
'dtx_controller.h',
'dtx_controller.cc',
'fec_controller.h',
@@ -33,32 +31,7 @@
'include/audio_network_adaptor.h',
'smoothing_filter.h',
'smoothing_filter.cc',
- ], # sources
- 'conditions': [
- ['enable_protobuf==1', {
- 'dependencies': ['debug_dump_proto'],
- 'defines': ['WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP'],
- }],
- ], # conditions
+ ], # source
},
], # targets
-
- 'conditions': [
- ['enable_protobuf==1', {
- 'targets': [
- { 'target_name': 'debug_dump_proto',
- 'type': 'static_library',
- 'sources': ['debug_dump.proto',],
- 'variables': {
- 'proto_in_dir': '.',
- # Workaround to protect against gyp's pathname relativization when
- # this file is included by modules.gyp.
- 'proto_out_protected': 'webrtc/modules/audio_coding/audio_network_adaptor',
- 'proto_out_dir': '<(proto_out_protected)',
- },
- 'includes': ['../../../build/protoc.gypi',],
- },
- ], # targets
- }],
- ], # conditions
}
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
index 3d8b2be..0303c84 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
@@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
-
#include <utility>
+#include "webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
+
namespace webrtc {
AudioNetworkAdaptorImpl::Config::Config() = default;
@@ -21,15 +21,7 @@
AudioNetworkAdaptorImpl::AudioNetworkAdaptorImpl(
const Config& config,
std::unique_ptr<ControllerManager> controller_manager)
- : AudioNetworkAdaptorImpl(config, std::move(controller_manager), nullptr) {}
-
-AudioNetworkAdaptorImpl::AudioNetworkAdaptorImpl(
- const Config& config,
- std::unique_ptr<ControllerManager> controller_manager,
- std::unique_ptr<DebugDumpWriter> debug_dump_writer)
- : config_(config),
- controller_manager_(std::move(controller_manager)),
- debug_dump_writer_(std::move(debug_dump_writer)) {
+ : config_(config), controller_manager_(std::move(controller_manager)) {
RTC_DCHECK(controller_manager_);
}
@@ -37,19 +29,16 @@
void AudioNetworkAdaptorImpl::SetUplinkBandwidth(int uplink_bandwidth_bps) {
last_metrics_.uplink_bandwidth_bps = rtc::Optional<int>(uplink_bandwidth_bps);
- DumpNetworkMetrics();
+
+ // TODO(minyue): Add debug dumping.
}
void AudioNetworkAdaptorImpl::SetUplinkPacketLossFraction(
float uplink_packet_loss_fraction) {
last_metrics_.uplink_packet_loss_fraction =
rtc::Optional<float>(uplink_packet_loss_fraction);
- DumpNetworkMetrics();
-}
-void AudioNetworkAdaptorImpl::SetRtt(int rtt_ms) {
- last_metrics_.rtt_ms = rtc::Optional<int>(rtt_ms);
- DumpNetworkMetrics();
+ // TODO(minyue): Add debug dumping.
}
AudioNetworkAdaptor::EncoderRuntimeConfig
@@ -60,9 +49,6 @@
controller->MakeDecision(last_metrics_, &config);
// TODO(minyue): Add debug dumping.
- if (debug_dump_writer_)
- debug_dump_writer_->DumpEncoderRuntimeConfig(
- config, config_.clock->TimeInMilliseconds());
return config;
}
@@ -81,17 +67,7 @@
}
void AudioNetworkAdaptorImpl::StartDebugDump(FILE* file_handle) {
- debug_dump_writer_ = DebugDumpWriter::Create(file_handle);
-}
-
-void AudioNetworkAdaptorImpl::StopDebugDump() {
- debug_dump_writer_.reset(nullptr);
-}
-
-void AudioNetworkAdaptorImpl::DumpNetworkMetrics() {
- if (debug_dump_writer_)
- debug_dump_writer_->DumpNetworkMetrics(last_metrics_,
- config_.clock->TimeInMilliseconds());
+ // TODO(minyue): Implement this method.
}
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
index 289b677..6f8d348 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
@@ -16,9 +16,7 @@
#include "webrtc/base/constructormagic.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/controller.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
-#include "webrtc/system_wrappers/include/clock.h"
namespace webrtc {
@@ -27,27 +25,18 @@
struct Config {
Config();
~Config();
- const Clock* clock;
};
AudioNetworkAdaptorImpl(
const Config& config,
std::unique_ptr<ControllerManager> controller_manager);
- // Dependency injection for testing.
- AudioNetworkAdaptorImpl(
- const Config& config,
- std::unique_ptr<ControllerManager> controller_manager,
- std::unique_ptr<DebugDumpWriter> debug_dump_writer = nullptr);
-
~AudioNetworkAdaptorImpl() override;
void SetUplinkBandwidth(int uplink_bandwidth_bps) override;
void SetUplinkPacketLossFraction(float uplink_packet_loss_fraction) override;
- void SetRtt(int rtt_ms) override;
-
void SetReceiverFrameLengthRange(int min_frame_length_ms,
int max_frame_length_ms) override;
@@ -55,17 +44,11 @@
void StartDebugDump(FILE* file_handle) override;
- void StopDebugDump() override;
-
private:
- void DumpNetworkMetrics();
-
const Config config_;
std::unique_ptr<ControllerManager> controller_manager_;
- std::unique_ptr<DebugDumpWriter> debug_dump_writer_;
-
Controller::NetworkMetrics last_metrics_;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioNetworkAdaptorImpl);
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
index aee41da..6562674 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
@@ -15,25 +15,20 @@
#include "webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller_manager.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h"
namespace webrtc {
using ::testing::_;
using ::testing::NiceMock;
using ::testing::Return;
-using ::testing::SetArgPointee;
namespace {
constexpr size_t kNumControllers = 2;
-constexpr int64_t kClockInitialTimeMs = 12345678;
-
MATCHER_P(NetworkMetricsIs, metric, "") {
return arg.uplink_bandwidth_bps == metric.uplink_bandwidth_bps &&
arg.target_audio_bitrate_bps == metric.target_audio_bitrate_bps &&
- arg.rtt_ms == metric.rtt_ms &&
arg.uplink_packet_loss_fraction == metric.uplink_packet_loss_fraction;
}
@@ -44,21 +39,9 @@
frame_length_range.max_frame_length_ms;
}
-MATCHER_P(EncoderRuntimeConfigIs, config, "") {
- return arg.bitrate_bps == config.bitrate_bps &&
- arg.frame_length_ms == config.frame_length_ms &&
- arg.uplink_packet_loss_fraction ==
- config.uplink_packet_loss_fraction &&
- arg.enable_fec == config.enable_fec &&
- arg.enable_dtx == config.enable_dtx &&
- arg.num_channels == config.num_channels;
-}
-
struct AudioNetworkAdaptorStates {
std::unique_ptr<AudioNetworkAdaptorImpl> audio_network_adaptor;
std::vector<std::unique_ptr<MockController>> mock_controllers;
- std::unique_ptr<SimulatedClock> simulated_clock;
- MockDebugDumpWriter* mock_debug_dump_writer;
};
AudioNetworkAdaptorStates CreateAudioNetworkAdaptor() {
@@ -81,19 +64,9 @@
EXPECT_CALL(*controller_manager, GetSortedControllers(_))
.WillRepeatedly(Return(controllers));
- states.simulated_clock.reset(new SimulatedClock(kClockInitialTimeMs * 1000));
-
- auto debug_dump_writer =
- std::unique_ptr<MockDebugDumpWriter>(new NiceMock<MockDebugDumpWriter>());
- EXPECT_CALL(*debug_dump_writer, Die());
- states.mock_debug_dump_writer = debug_dump_writer.get();
-
- AudioNetworkAdaptorImpl::Config config;
- config.clock = states.simulated_clock.get();
// AudioNetworkAdaptorImpl governs the lifetime of controller manager.
states.audio_network_adaptor.reset(new AudioNetworkAdaptorImpl(
- config,
- std::move(controller_manager), std::move(debug_dump_writer)));
+ AudioNetworkAdaptorImpl::Config(), std::move(controller_manager)));
return states;
}
@@ -135,52 +108,4 @@
states.audio_network_adaptor->SetReceiverFrameLengthRange(20, 120);
}
-TEST(AudioNetworkAdaptorImplTest,
- DumpEncoderRuntimeConfigIsCalledOnGetEncoderRuntimeConfig) {
- auto states = CreateAudioNetworkAdaptor();
-
- AudioNetworkAdaptor::EncoderRuntimeConfig config;
- config.bitrate_bps = rtc::Optional<int>(32000);
- config.enable_fec = rtc::Optional<bool>(true);
-
- EXPECT_CALL(*states.mock_controllers[0], MakeDecision(_, _))
- .WillOnce(SetArgPointee<1>(config));
-
- EXPECT_CALL(*states.mock_debug_dump_writer,
- DumpEncoderRuntimeConfig(EncoderRuntimeConfigIs(config),
- kClockInitialTimeMs));
- states.audio_network_adaptor->GetEncoderRuntimeConfig();
-}
-
-TEST(AudioNetworkAdaptorImplTest,
- DumpNetworkMetricsIsCalledOnSetNetworkMetrics) {
- auto states = CreateAudioNetworkAdaptor();
-
- constexpr int kBandwidth = 16000;
- constexpr float kPacketLoss = 0.7f;
- constexpr int kRtt = 100;
-
- Controller::NetworkMetrics check;
- check.uplink_bandwidth_bps = rtc::Optional<int>(kBandwidth);
- int64_t timestamp_check = kClockInitialTimeMs;
-
- EXPECT_CALL(*states.mock_debug_dump_writer,
- DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
- states.audio_network_adaptor->SetUplinkBandwidth(kBandwidth);
-
- states.simulated_clock->AdvanceTimeMilliseconds(100);
- timestamp_check += 100;
- check.uplink_packet_loss_fraction = rtc::Optional<float>(kPacketLoss);
- EXPECT_CALL(*states.mock_debug_dump_writer,
- DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
- states.audio_network_adaptor->SetUplinkPacketLossFraction(kPacketLoss);
-
- states.simulated_clock->AdvanceTimeMilliseconds(200);
- timestamp_check += 200;
- check.rtt_ms = rtc::Optional<int>(kRtt);
- EXPECT_CALL(*states.mock_debug_dump_writer,
- DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
- states.audio_network_adaptor->SetRtt(kRtt);
-}
-
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/controller.h b/webrtc/modules/audio_coding/audio_network_adaptor/controller.h
index f27a391..c1b16c7 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/controller.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/controller.h
@@ -24,7 +24,6 @@
rtc::Optional<int> uplink_bandwidth_bps;
rtc::Optional<float> uplink_packet_loss_fraction;
rtc::Optional<int> target_audio_bitrate_bps;
- rtc::Optional<int> rtt_ms;
};
struct Constraints {
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto
deleted file mode 100644
index f425244..0000000
--- a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto
+++ /dev/null
@@ -1,30 +0,0 @@
-syntax = "proto2";
-option optimize_for = LITE_RUNTIME;
-package webrtc.audio_network_adaptor.debug_dump;
-
-message NetworkMetrics {
- optional int32 uplink_bandwidth_bps = 1;
- optional float uplink_packet_loss_fraction = 2;
- optional int32 target_audio_bitrate_bps = 3;
- optional int32 rtt_ms = 4;
-}
-
-message EncoderRuntimeConfig {
- optional int32 bitrate_bps = 1;
- optional int32 frame_length_ms = 2;
- optional float uplink_packet_loss_fraction = 3;
- optional bool enable_fec = 4;
- optional bool enable_dtx = 5;
- optional uint32 num_channels = 6;
-}
-
-message Event {
- enum Type {
- NETWORK_METRICS = 0;
- ENCODER_RUNTIME_CONFIG = 1;
- }
- required Type type = 1;
- required uint32 timestamp = 2;
- optional NetworkMetrics network_metrics = 3;
- optional EncoderRuntimeConfig encoder_runtime_config = 4;
-}
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
deleted file mode 100644
index 9992e2d..0000000
--- a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
+++ /dev/null
@@ -1,135 +0,0 @@
-/*
- * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
-
-#include "webrtc/base/checks.h"
-
-#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
-#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
-#include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.pb.h"
-#else
-#include "webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.pb.h"
-#endif
-#endif
-
-namespace webrtc {
-
-#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
-namespace {
-
-using audio_network_adaptor::debug_dump::Event;
-using audio_network_adaptor::debug_dump::NetworkMetrics;
-using audio_network_adaptor::debug_dump::EncoderRuntimeConfig;
-
-void DumpEventToFile(const Event& event, FileWrapper* dump_file) {
- RTC_CHECK(dump_file->is_open());
- std::string dump_data;
- event.SerializeToString(&dump_data);
- int32_t size = event.ByteSize();
- dump_file->Write(&size, sizeof(size));
- dump_file->Write(dump_data.data(), dump_data.length());
-}
-
-} // namespace
-#endif // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
-
-class DebugDumpWriterImpl final : public DebugDumpWriter {
- public:
- explicit DebugDumpWriterImpl(FILE* file_handle);
- ~DebugDumpWriterImpl() override = default;
-
- void DumpEncoderRuntimeConfig(
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config,
- int64_t timestamp) override;
-
- void DumpNetworkMetrics(const Controller::NetworkMetrics& metrics,
- int64_t timestamp) override;
-
- private:
- std::unique_ptr<FileWrapper> dump_file_;
-};
-
-DebugDumpWriterImpl::DebugDumpWriterImpl(FILE* file_handle)
- : dump_file_(FileWrapper::Create()) {
-#ifndef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
- RTC_DCHECK(false);
-#endif
- dump_file_->OpenFromFileHandle(file_handle);
- RTC_CHECK(dump_file_->is_open());
-}
-
-void DebugDumpWriterImpl::DumpNetworkMetrics(
- const Controller::NetworkMetrics& metrics,
- int64_t timestamp) {
-#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
- Event event;
- event.set_timestamp(timestamp);
- event.set_type(Event::NETWORK_METRICS);
- auto dump_metrics = event.mutable_network_metrics();
-
- if (metrics.uplink_bandwidth_bps)
- dump_metrics->set_uplink_bandwidth_bps(*metrics.uplink_bandwidth_bps);
-
- if (metrics.uplink_packet_loss_fraction) {
- dump_metrics->set_uplink_packet_loss_fraction(
- *metrics.uplink_packet_loss_fraction);
- }
-
- if (metrics.target_audio_bitrate_bps) {
- dump_metrics->set_target_audio_bitrate_bps(
- *metrics.target_audio_bitrate_bps);
- }
-
- if (metrics.rtt_ms)
- dump_metrics->set_rtt_ms(*metrics.rtt_ms);
-
- DumpEventToFile(event, dump_file_.get());
-#endif // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
-}
-
-void DebugDumpWriterImpl::DumpEncoderRuntimeConfig(
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config,
- int64_t timestamp) {
-#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
- Event event;
- event.set_timestamp(timestamp);
- event.set_type(Event::ENCODER_RUNTIME_CONFIG);
- auto dump_config = event.mutable_encoder_runtime_config();
-
- if (config.bitrate_bps)
- dump_config->set_bitrate_bps(*config.bitrate_bps);
-
- if (config.frame_length_ms)
- dump_config->set_frame_length_ms(*config.frame_length_ms);
-
- if (config.uplink_packet_loss_fraction) {
- dump_config->set_uplink_packet_loss_fraction(
- *config.uplink_packet_loss_fraction);
- }
-
- if (config.enable_fec)
- dump_config->set_enable_fec(*config.enable_fec);
-
- if (config.enable_dtx)
- dump_config->set_enable_dtx(*config.enable_dtx);
-
- if (config.num_channels)
- dump_config->set_num_channels(*config.num_channels);
-
- DumpEventToFile(event, dump_file_.get());
-#endif // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
-}
-
-std::unique_ptr<DebugDumpWriter> DebugDumpWriter::Create(FILE* file_handle) {
- return std::unique_ptr<DebugDumpWriter>(new DebugDumpWriterImpl(file_handle));
-}
-
-} // namespace webrtc
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
deleted file mode 100644
index da4b031..0000000
--- a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
- * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP_WRITER_H_
-#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"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
-#include "webrtc/system_wrappers/include/file_wrapper.h"
-
-namespace webrtc {
-
-class DebugDumpWriter {
- public:
- static std::unique_ptr<DebugDumpWriter> Create(FILE* file_handle);
-
- virtual ~DebugDumpWriter() = default;
-
- virtual void DumpEncoderRuntimeConfig(
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config,
- int64_t timestamp) = 0;
-
- virtual void DumpNetworkMetrics(const Controller::NetworkMetrics& metrics,
- int64_t timestamp) = 0;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP_WRITER_H_
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h b/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
index e3c4db0..4b43276 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
@@ -43,16 +43,12 @@
virtual void SetUplinkPacketLossFraction(
float uplink_packet_loss_fraction) = 0;
- virtual void SetRtt(int rtt_ms) = 0;
-
virtual void SetReceiverFrameLengthRange(int min_frame_length_ms,
int max_frame_length_ms) = 0;
virtual EncoderRuntimeConfig GetEncoderRuntimeConfig() = 0;
virtual void StartDebugDump(FILE* file_handle) = 0;
-
- virtual void StopDebugDump() = 0;
};
} // 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
deleted file mode 100644
index f7e226e..0000000
--- a/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_DEBUG_DUMP_WRITER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_DEBUG_DUMP_WRITER_H_
-
-#include "testing/gmock/include/gmock/gmock.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
-
-namespace webrtc {
-
-class MockDebugDumpWriter : public DebugDumpWriter {
- public:
- virtual ~MockDebugDumpWriter() { Die(); }
- MOCK_METHOD0(Die, void());
-
- MOCK_METHOD2(DumpEncoderRuntimeConfig,
- void(const AudioNetworkAdaptor::EncoderRuntimeConfig& config,
- int64_t timestamp));
- MOCK_METHOD2(DumpNetworkMetrics,
- void(const Controller::NetworkMetrics& metrics,
- int64_t timestamp));
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_DEBUG_DUMP_WRITER_H_