Relanding of "Adding debug dump to audio network adaptor."
The original CL was https://codereview.webrtc.org/2356763002
but got reverted https://codereview.webrtc.org/2362003002/.
The error was that ana_debug_dump_proto as a proto_library was placed under rtc_include_tests.
BUG=webrtc:6303
Review-Url: https://codereview.webrtc.org/2365723002
Cr-Commit-Position: refs/heads/master@{#14363}
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index eaf2e74..a69206e 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -696,6 +696,15 @@
}
}
+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"
+ }
+}
+
source_set("audio_network_adaptor") {
sources = [
"audio_network_adaptor/audio_network_adaptor.cc",
@@ -709,6 +718,8 @@
"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",
@@ -721,6 +732,13 @@
]
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") {
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 af6ba55..508b62b 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
@@ -22,6 +22,8 @@
'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',
@@ -31,7 +33,32 @@
'include/audio_network_adaptor.h',
'smoothing_filter.h',
'smoothing_filter.cc',
- ], # source
+ ], # sources
+ 'conditions': [
+ ['enable_protobuf==1', {
+ 'dependencies': ['debug_dump_proto'],
+ 'defines': ['WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP'],
+ }],
+ ], # conditions
},
], # 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 0303c84..3d8b2be 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 <utility>
-
#include "webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
+#include <utility>
+
namespace webrtc {
AudioNetworkAdaptorImpl::Config::Config() = default;
@@ -21,7 +21,15 @@
AudioNetworkAdaptorImpl::AudioNetworkAdaptorImpl(
const Config& config,
std::unique_ptr<ControllerManager> controller_manager)
- : config_(config), controller_manager_(std::move(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)) {
RTC_DCHECK(controller_manager_);
}
@@ -29,16 +37,19 @@
void AudioNetworkAdaptorImpl::SetUplinkBandwidth(int uplink_bandwidth_bps) {
last_metrics_.uplink_bandwidth_bps = rtc::Optional<int>(uplink_bandwidth_bps);
-
- // TODO(minyue): Add debug dumping.
+ DumpNetworkMetrics();
}
void AudioNetworkAdaptorImpl::SetUplinkPacketLossFraction(
float uplink_packet_loss_fraction) {
last_metrics_.uplink_packet_loss_fraction =
rtc::Optional<float>(uplink_packet_loss_fraction);
+ DumpNetworkMetrics();
+}
- // TODO(minyue): Add debug dumping.
+void AudioNetworkAdaptorImpl::SetRtt(int rtt_ms) {
+ last_metrics_.rtt_ms = rtc::Optional<int>(rtt_ms);
+ DumpNetworkMetrics();
}
AudioNetworkAdaptor::EncoderRuntimeConfig
@@ -49,6 +60,9 @@
controller->MakeDecision(last_metrics_, &config);
// TODO(minyue): Add debug dumping.
+ if (debug_dump_writer_)
+ debug_dump_writer_->DumpEncoderRuntimeConfig(
+ config, config_.clock->TimeInMilliseconds());
return config;
}
@@ -67,7 +81,17 @@
}
void AudioNetworkAdaptorImpl::StartDebugDump(FILE* file_handle) {
- // TODO(minyue): Implement this method.
+ 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());
}
} // 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 6f8d348..289b677 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
@@ -16,7 +16,9 @@
#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 {
@@ -25,18 +27,27 @@
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;
@@ -44,11 +55,17 @@
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 6562674..aee41da 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
@@ -15,20 +15,25 @@
#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;
}
@@ -39,9 +44,21 @@
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() {
@@ -64,9 +81,19 @@
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(
- AudioNetworkAdaptorImpl::Config(), std::move(controller_manager)));
+ config,
+ std::move(controller_manager), std::move(debug_dump_writer)));
return states;
}
@@ -108,4 +135,52 @@
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 c1b16c7..f27a391 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/controller.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/controller.h
@@ -24,6 +24,7 @@
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
new file mode 100644
index 0000000..f425244
--- /dev/null
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto
@@ -0,0 +1,30 @@
+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
new file mode 100644
index 0000000..9992e2d
--- /dev/null
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
@@ -0,0 +1,135 @@
+/*
+ * 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
new file mode 100644
index 0000000..da4b031
--- /dev/null
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h
@@ -0,0 +1,40 @@
+/*
+ * 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 4b43276..e3c4db0 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
@@ -43,12 +43,16 @@
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
new file mode 100644
index 0000000..f7e226e
--- /dev/null
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h
@@ -0,0 +1,34 @@
+/*
+ * 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_
