Revert of Log audio network adapter decisions in event log. (patchset #14 id:320001 of https://codereview.webrtc.org/2559953002/ )
Reason for revert:
Breaks chromium.webrtc.fyi.
Original issue's description:
> Log audio network adapter decisions in event log.
>
> BUG=webrtc:6845
>
> Review-Url: https://codereview.webrtc.org/2559953002
> Cr-Commit-Position: refs/heads/master@{#16053}
> Committed: https://chromium.googlesource.com/external/webrtc/+/3663681b5d05682f38e75d9eaf6049299f62fb02
TBR=minyue@webrtc.org,henrik.lundin@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,solenberg@webrtc.org,michaelt@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6845
Review-Url: https://codereview.webrtc.org/2631703002
Cr-Commit-Position: refs/heads/master@{#16054}
diff --git a/webrtc/logging/BUILD.gn b/webrtc/logging/BUILD.gn
index 8bce39a..2daaef4 100644
--- a/webrtc/logging/BUILD.gn
+++ b/webrtc/logging/BUILD.gn
@@ -42,7 +42,6 @@
":rtc_event_log_api",
"..:webrtc_common",
"../call:call_interfaces",
- "../modules/audio_coding:audio_network_adaptor",
"../modules/rtp_rtcp",
]
@@ -85,8 +84,7 @@
sources = [
"rtc_event_log/rtc_event_log.proto",
]
- proto_in_dir = "//"
- proto_out_dir = "."
+ proto_out_dir = "webrtc/logging/rtc_event_log"
}
}
diff --git a/webrtc/logging/rtc_event_log/DEPS b/webrtc/logging/rtc_event_log/DEPS
index e340150..39d2020 100644
--- a/webrtc/logging/rtc_event_log/DEPS
+++ b/webrtc/logging/rtc_event_log/DEPS
@@ -1,7 +1,6 @@
include_rules = [
"+webrtc/base",
"+webrtc/call",
- "+webrtc/modules/audio_coding/audio_network_adaptor",
"+webrtc/modules/rtp_rtcp",
"+webrtc/system_wrappers",
]
diff --git a/webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h b/webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h
index 4db33a1..315967b 100644
--- a/webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h
+++ b/webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h
@@ -58,8 +58,6 @@
void(int32_t bitrate,
uint8_t fraction_loss,
int32_t total_packets));
- MOCK_METHOD1(LogAudioNetworkAdaptation,
- void(const AudioNetworkAdaptor::EncoderRuntimeConfig& config));
};
} // namespace webrtc
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log.cc b/webrtc/logging/rtc_event_log/rtc_event_log.cc
index 65ee7d8..21ca5e1 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log.cc
+++ b/webrtc/logging/rtc_event_log/rtc_event_log.cc
@@ -77,8 +77,6 @@
void LogBwePacketLossEvent(int32_t bitrate,
uint8_t fraction_loss,
int32_t total_packets) override;
- void LogAudioNetworkAdaptation(
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config) override;
private:
void StoreEvent(std::unique_ptr<rtclog::Event>* event);
@@ -436,29 +434,6 @@
StoreEvent(&event);
}
-void RtcEventLogImpl::LogAudioNetworkAdaptation(
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config) {
- std::unique_ptr<rtclog::Event> event(new rtclog::Event());
- event->set_timestamp_us(rtc::TimeMicros());
- event->set_type(rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT);
- auto audio_network_adaptation = event->mutable_audio_network_adaptation();
- if (config.bitrate_bps)
- audio_network_adaptation->set_bitrate_bps(*config.bitrate_bps);
- if (config.frame_length_ms)
- audio_network_adaptation->set_frame_length_ms(*config.frame_length_ms);
- if (config.uplink_packet_loss_fraction) {
- audio_network_adaptation->set_uplink_packet_loss_fraction(
- *config.uplink_packet_loss_fraction);
- }
- if (config.enable_fec)
- audio_network_adaptation->set_enable_fec(*config.enable_fec);
- if (config.enable_dtx)
- audio_network_adaptation->set_enable_dtx(*config.enable_dtx);
- if (config.num_channels)
- audio_network_adaptation->set_num_channels(*config.num_channels);
- StoreEvent(&event);
-}
-
void RtcEventLogImpl::StoreEvent(std::unique_ptr<rtclog::Event>* event) {
if (!event_queue_.Insert(event)) {
LOG(LS_ERROR) << "WebRTC event log queue full. Dropping event.";
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log.h b/webrtc/logging/rtc_event_log/rtc_event_log.h
index 5d221d4..ccf6094 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log.h
+++ b/webrtc/logging/rtc_event_log/rtc_event_log.h
@@ -17,7 +17,6 @@
#include "webrtc/base/platform_file.h"
#include "webrtc/call/audio_receive_stream.h"
#include "webrtc/call/audio_send_stream.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
#include "webrtc/video_receive_stream.h"
#include "webrtc/video_send_stream.h"
@@ -115,10 +114,6 @@
uint8_t fraction_loss,
int32_t total_packets) = 0;
- // Logs audio encoder re-configuration driven by audio network adaptor.
- virtual void LogAudioNetworkAdaptation(
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config) = 0;
-
// Reads an RtcEventLog file and returns true when reading was successful.
// The result is stored in the given EventStream object.
// The order of the events in the EventStream is implementation defined.
@@ -160,8 +155,6 @@
void LogBwePacketLossEvent(int32_t bitrate,
uint8_t fraction_loss,
int32_t total_packets) override {}
- void LogAudioNetworkAdaptation(
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config) override{};
};
} // namespace webrtc
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log.proto b/webrtc/logging/rtc_event_log/rtc_event_log.proto
index e807722..a6d1695 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log.proto
+++ b/webrtc/logging/rtc_event_log/rtc_event_log.proto
@@ -37,7 +37,6 @@
VIDEO_SENDER_CONFIG_EVENT = 9;
AUDIO_RECEIVER_CONFIG_EVENT = 10;
AUDIO_SENDER_CONFIG_EVENT = 11;
- AUDIO_NETWORK_ADAPTATION_EVENT = 16;
}
// required - Indicates the type of this event
@@ -66,9 +65,6 @@
// optional - but required if type == AUDIO_SENDER_CONFIG_EVENT
optional AudioSendConfig audio_sender_config = 11;
-
- // optional - but required if type == AUDIO_NETWORK_ADAPTATION_EVENT
- optional AudioNetworkAdaptation audio_network_adaptation = 16;
}
message RtpPacket {
@@ -231,24 +227,3 @@
// RTP header extensions used for the outgoing audio stream.
repeated RtpHeaderExtension header_extensions = 2;
}
-
-message AudioNetworkAdaptation {
- // Bit rate that the audio encoder is operating at.
- optional int32 bitrate_bps = 1;
-
- // Frame length that each encoded audio packet consists of.
- optional int32 frame_length_ms = 2;
-
- // Packet loss fraction that the encoder's forward error correction (FEC) is
- // optimized for.
- optional float uplink_packet_loss_fraction = 3;
-
- // Whether forward error correction (FEC) is turned on or off.
- optional bool enable_fec = 4;
-
- // Whether discontinuous transmission (DTX) is turned on or off.
- optional bool enable_dtx = 5;
-
- // Number of audio channels that each encoded packet consists of.
- optional uint32 num_channels = 6;
-}
\ No newline at end of file
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc b/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
index 3b808b2..ce55a4f 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
+++ b/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
@@ -79,8 +79,6 @@
return ParsedRtcEventLog::EventType::AUDIO_RECEIVER_CONFIG_EVENT;
case rtclog::Event::AUDIO_SENDER_CONFIG_EVENT:
return ParsedRtcEventLog::EventType::AUDIO_SENDER_CONFIG_EVENT;
- case rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT:
- return ParsedRtcEventLog::EventType::AUDIO_NETWORK_ADAPTATION_EVENT;
}
RTC_NOTREACHED();
return ParsedRtcEventLog::EventType::UNKNOWN_EVENT;
@@ -456,29 +454,4 @@
}
}
-void ParsedRtcEventLog::GetAudioNetworkAdaptation(
- size_t index,
- AudioNetworkAdaptor::EncoderRuntimeConfig* config) const {
- RTC_CHECK_LT(index, GetNumberOfEvents());
- const rtclog::Event& event = events_[index];
- RTC_CHECK(event.has_type());
- RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT);
- RTC_CHECK(event.has_audio_network_adaptation());
- const rtclog::AudioNetworkAdaptation& ana_event =
- event.audio_network_adaptation();
- if (ana_event.has_bitrate_bps())
- config->bitrate_bps = rtc::Optional<int>(ana_event.bitrate_bps());
- if (ana_event.has_enable_fec())
- config->enable_fec = rtc::Optional<bool>(ana_event.enable_fec());
- if (ana_event.has_enable_dtx())
- config->enable_dtx = rtc::Optional<bool>(ana_event.enable_dtx());
- if (ana_event.has_frame_length_ms())
- config->frame_length_ms = rtc::Optional<int>(ana_event.frame_length_ms());
- if (ana_event.has_num_channels())
- config->num_channels = rtc::Optional<size_t>(ana_event.num_channels());
- if (ana_event.has_uplink_packet_loss_fraction())
- config->uplink_packet_loss_fraction =
- rtc::Optional<float>(ana_event.uplink_packet_loss_fraction());
-}
-
} // namespace webrtc
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_parser.h b/webrtc/logging/rtc_event_log/rtc_event_log_parser.h
index 8472668..2d66b90 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log_parser.h
+++ b/webrtc/logging/rtc_event_log/rtc_event_log_parser.h
@@ -47,8 +47,7 @@
VIDEO_RECEIVER_CONFIG_EVENT = 8,
VIDEO_SENDER_CONFIG_EVENT = 9,
AUDIO_RECEIVER_CONFIG_EVENT = 10,
- AUDIO_SENDER_CONFIG_EVENT = 11,
- AUDIO_NETWORK_ADAPTATION_EVENT = 16
+ AUDIO_SENDER_CONFIG_EVENT = 11
};
// Reads an RtcEventLog file and returns true if parsing was successful.
@@ -124,13 +123,6 @@
uint8_t* fraction_loss,
int32_t* total_packets) const;
- // Reads a audio network adaptation event to a (non-NULL)
- // AudioNetworkAdaptor::EncoderRuntimeConfig struct. Only the fields that are
- // stored in the protobuf will be written.
- void GetAudioNetworkAdaptation(
- size_t index,
- AudioNetworkAdaptor::EncoderRuntimeConfig* config) const;
-
private:
std::vector<rtclog::Event> events_;
};
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc b/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
index f0ee973..1540466 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
+++ b/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
@@ -228,19 +228,6 @@
}
}
-void GenerateAudioNetworkAdaptation(
- uint32_t extensions_bitvector,
- AudioNetworkAdaptor::EncoderRuntimeConfig* config,
- Random* prng) {
- config->bitrate_bps = rtc::Optional<int>(prng->Rand(0, 3000000));
- config->enable_fec = rtc::Optional<bool>(prng->Rand<bool>());
- config->enable_dtx = rtc::Optional<bool>(prng->Rand<bool>());
- config->frame_length_ms = rtc::Optional<int>(prng->Rand(10, 120));
- config->num_channels = rtc::Optional<size_t>(prng->Rand(1, 2));
- config->uplink_packet_loss_fraction =
- rtc::Optional<float>(prng->Rand<float>());
-}
-
// Test for the RtcEventLog class. Dumps some RTP packets and other events
// to disk, then reads them back to see if they match.
void LogSessionAndReadBack(size_t rtp_count,
@@ -617,22 +604,6 @@
VideoSendStream::Config config;
};
-class AudioNetworkAdaptationReadWriteTest : public ConfigReadWriteTest {
- public:
- void GenerateConfig(uint32_t extensions_bitvector) override {
- GenerateAudioNetworkAdaptation(extensions_bitvector, &config, &prng);
- }
- void LogConfig(RtcEventLog* event_log) override {
- event_log->LogAudioNetworkAdaptation(config);
- }
- void VerifyConfig(const ParsedRtcEventLog& parsed_log,
- size_t index) override {
- RtcEventLogTestHelper::VerifyAudioNetworkAdaptation(parsed_log, index,
- config);
- }
- AudioNetworkAdaptor::EncoderRuntimeConfig config;
-};
-
TEST(RtcEventLogTest, LogAudioReceiveConfig) {
AudioReceiveConfigReadWriteTest test;
test.DoTest();
@@ -652,10 +623,4 @@
VideoSendConfigReadWriteTest test;
test.DoTest();
}
-
-TEST(RtcEventLogTest, LogAudioNetworkAdaptation) {
- AudioNetworkAdaptationReadWriteTest test;
- test.DoTest();
-}
-
} // namespace webrtc
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc b/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
index 19ca8aa..fb60dbc 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
+++ b/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
@@ -461,21 +461,6 @@
EXPECT_EQ(total_packets, parsed_total_packets);
}
-void RtcEventLogTestHelper::VerifyAudioNetworkAdaptation(
- const ParsedRtcEventLog& parsed_log,
- size_t index,
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config) {
- AudioNetworkAdaptor::EncoderRuntimeConfig parsed_config;
- parsed_log.GetAudioNetworkAdaptation(index, &parsed_config);
- EXPECT_EQ(config.bitrate_bps, parsed_config.bitrate_bps);
- EXPECT_EQ(config.enable_dtx, parsed_config.enable_dtx);
- EXPECT_EQ(config.enable_fec, parsed_config.enable_fec);
- EXPECT_EQ(config.frame_length_ms, parsed_config.frame_length_ms);
- EXPECT_EQ(config.num_channels, parsed_config.num_channels);
- EXPECT_EQ(config.uplink_packet_loss_fraction,
- parsed_config.uplink_packet_loss_fraction);
-}
-
void RtcEventLogTestHelper::VerifyLogStartEvent(
const ParsedRtcEventLog& parsed_log,
size_t index) {
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h b/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h
index 2f4e177..3f89e30 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h
+++ b/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h
@@ -56,11 +56,6 @@
uint8_t fraction_loss,
int32_t total_packets);
- static void VerifyAudioNetworkAdaptation(
- const ParsedRtcEventLog& parsed_log,
- size_t index,
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config);
-
static void VerifyLogStartEvent(const ParsedRtcEventLog& parsed_log,
size_t index);
static void VerifyLogEndEvent(const ParsedRtcEventLog& parsed_log,
diff --git a/webrtc/modules/BUILD.gn b/webrtc/modules/BUILD.gn
index 2b941a8..f3b2ac9 100644
--- a/webrtc/modules/BUILD.gn
+++ b/webrtc/modules/BUILD.gn
@@ -258,7 +258,6 @@
"audio_coding/audio_network_adaptor/channel_controller_unittest.cc",
"audio_coding/audio_network_adaptor/controller_manager_unittest.cc",
"audio_coding/audio_network_adaptor/dtx_controller_unittest.cc",
- "audio_coding/audio_network_adaptor/event_log_writer_unittest.cc",
"audio_coding/audio_network_adaptor/fec_controller_unittest.cc",
"audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc",
"audio_coding/audio_network_adaptor/mock/mock_controller.h",
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index a8799be..c38cd47 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -887,8 +887,7 @@
sources = [
"audio_network_adaptor/debug_dump.proto",
]
- proto_in_dir = "//"
- proto_out_dir = "."
+ proto_out_dir = "webrtc/modules/audio_coding/audio_network_adaptor"
}
proto_library("ana_config_proto") {
sources = [
@@ -915,8 +914,6 @@
"audio_network_adaptor/debug_dump_writer.h",
"audio_network_adaptor/dtx_controller.cc",
"audio_network_adaptor/dtx_controller.h",
- "audio_network_adaptor/event_log_writer.cc",
- "audio_network_adaptor/event_log_writer.h",
"audio_network_adaptor/fec_controller.cc",
"audio_network_adaptor/fec_controller.h",
"audio_network_adaptor/frame_length_controller.cc",
@@ -928,7 +925,6 @@
"../..:webrtc_common",
"../../base:rtc_base_approved",
"../../common_audio",
- "../../logging:rtc_event_log_api",
"../../system_wrappers",
]
@@ -939,11 +935,6 @@
]
defines = [ "WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP" ]
}
-
- 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" ]
- }
}
rtc_static_library("neteq") {
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 6ec92b0..3b767d9 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
@@ -12,16 +12,8 @@
#include <utility>
-#include "webrtc/base/logging.h"
-
namespace webrtc {
-namespace {
-constexpr int kEventLogMinBitrateChangeBps = 5000;
-constexpr float kEventLogMinBitrateChangeFraction = 0.25;
-constexpr float kEventLogMinPacketLossChangeFraction = 0.5;
-} // namespace
-
AudioNetworkAdaptorImpl::Config::Config()
: event_log(nullptr), clock(nullptr){};
@@ -33,14 +25,7 @@
std::unique_ptr<DebugDumpWriter> debug_dump_writer)
: config_(config),
controller_manager_(std::move(controller_manager)),
- debug_dump_writer_(std::move(debug_dump_writer)),
- event_log_writer_(
- config.event_log
- ? new EventLogWriter(config.event_log,
- kEventLogMinBitrateChangeBps,
- kEventLogMinBitrateChangeFraction,
- kEventLogMinPacketLossChangeFraction)
- : nullptr) {
+ debug_dump_writer_(std::move(debug_dump_writer)) {
RTC_DCHECK(controller_manager_);
}
@@ -83,13 +68,11 @@
controller_manager_->GetSortedControllers(last_metrics_))
controller->MakeDecision(last_metrics_, &config);
+ // TODO(minyue): Add debug dumping.
if (debug_dump_writer_)
debug_dump_writer_->DumpEncoderRuntimeConfig(
config, config_.clock->TimeInMilliseconds());
- if (event_log_writer_)
- event_log_writer_->MaybeLogEncoderConfig(config);
-
return config;
}
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 9c47125..801a9cc 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
@@ -17,7 +17,6 @@
#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/event_log_writer.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
#include "webrtc/system_wrappers/include/clock.h"
@@ -66,8 +65,6 @@
std::unique_ptr<DebugDumpWriter> debug_dump_writer_;
- const std::unique_ptr<EventLogWriter> event_log_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 9b608dc..b26d2f5 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
@@ -11,7 +11,6 @@
#include <utility>
#include <vector>
-#include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h"
#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"
@@ -53,7 +52,6 @@
std::unique_ptr<AudioNetworkAdaptorImpl> audio_network_adaptor;
std::vector<std::unique_ptr<MockController>> mock_controllers;
std::unique_ptr<SimulatedClock> simulated_clock;
- std::unique_ptr<MockRtcEventLog> event_log;
MockDebugDumpWriter* mock_debug_dump_writer;
};
@@ -78,7 +76,6 @@
.WillRepeatedly(Return(controllers));
states.simulated_clock.reset(new SimulatedClock(kClockInitialTimeMs * 1000));
- states.event_log.reset(new NiceMock<MockRtcEventLog>());
auto debug_dump_writer =
std::unique_ptr<MockDebugDumpWriter>(new NiceMock<MockDebugDumpWriter>());
@@ -87,7 +84,6 @@
AudioNetworkAdaptorImpl::Config config;
config.clock = states.simulated_clock.get();
- config.event_log = states.event_log.get();
// AudioNetworkAdaptorImpl governs the lifetime of controller manager.
states.audio_network_adaptor.reset(new AudioNetworkAdaptorImpl(
config,
@@ -209,20 +205,4 @@
states.audio_network_adaptor->SetOverhead(kOverhead);
}
-TEST(AudioNetworkAdaptorImplTest, LogRuntimeConfigOnGetEncoderRuntimeConfig) {
- 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.event_log,
- LogAudioNetworkAdaptation(EncoderRuntimeConfigIs(config)))
- .Times(1);
- states.audio_network_adaptor->GetEncoderRuntimeConfig();
-}
-
} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc b/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
deleted file mode 100644
index 619a247..0000000
--- a/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
+++ /dev/null
@@ -1,68 +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 <math.h>
-#include <algorithm>
-
-#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.h"
-
-namespace webrtc {
-
-EventLogWriter::EventLogWriter(RtcEventLog* event_log,
- int min_bitrate_change_bps,
- float min_bitrate_change_fraction,
- float min_packet_loss_change_fraction)
- : event_log_(event_log),
- min_bitrate_change_bps_(min_bitrate_change_bps),
- min_bitrate_change_fraction_(min_bitrate_change_fraction),
- min_packet_loss_change_fraction_(min_packet_loss_change_fraction) {
- RTC_DCHECK(event_log_);
-}
-
-EventLogWriter::~EventLogWriter() = default;
-
-void EventLogWriter::MaybeLogEncoderConfig(
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config) {
- if (last_logged_config_.num_channels != config.num_channels)
- return LogEncoderConfig(config);
- if (last_logged_config_.enable_dtx != config.enable_dtx)
- return LogEncoderConfig(config);
- if (last_logged_config_.enable_fec != config.enable_fec)
- return LogEncoderConfig(config);
- if (last_logged_config_.frame_length_ms != config.frame_length_ms)
- return LogEncoderConfig(config);
- if ((!last_logged_config_.bitrate_bps && config.bitrate_bps) ||
- (last_logged_config_.bitrate_bps && config.bitrate_bps &&
- std::abs(*last_logged_config_.bitrate_bps - *config.bitrate_bps) >=
- std::min(static_cast<int>(*last_logged_config_.bitrate_bps *
- min_bitrate_change_fraction_),
- min_bitrate_change_bps_))) {
- return LogEncoderConfig(config);
- }
- if ((!last_logged_config_.uplink_packet_loss_fraction &&
- config.uplink_packet_loss_fraction) ||
- (last_logged_config_.uplink_packet_loss_fraction &&
- config.uplink_packet_loss_fraction &&
- fabs(*last_logged_config_.uplink_packet_loss_fraction -
- *config.uplink_packet_loss_fraction) >=
- min_packet_loss_change_fraction_ *
- *last_logged_config_.uplink_packet_loss_fraction)) {
- return LogEncoderConfig(config);
- }
-}
-
-void EventLogWriter::LogEncoderConfig(
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config) {
- event_log_->LogAudioNetworkAdaptation(config);
- last_logged_config_ = config;
-}
-
-} // namespace webrtc
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.h b/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.h
deleted file mode 100644
index 740da8c..0000000
--- a/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.h
+++ /dev/null
@@ -1,44 +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_EVENT_LOG_WRITER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_EVENT_LOG_WRITER_H_
-
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
-
-namespace webrtc {
-class RtcEventLog;
-
-class EventLogWriter final {
- public:
- EventLogWriter(RtcEventLog* event_log,
- int min_bitrate_change_bps,
- float min_bitrate_change_fraction,
- float min_packet_loss_change_fraction);
- ~EventLogWriter();
- void MaybeLogEncoderConfig(
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config);
-
- private:
- void LogEncoderConfig(
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config);
-
- RtcEventLog* const event_log_;
- const int min_bitrate_change_bps_;
- const float min_bitrate_change_fraction_;
- const float min_packet_loss_change_fraction_;
- AudioNetworkAdaptor::EncoderRuntimeConfig last_logged_config_;
- RTC_DISALLOW_COPY_AND_ASSIGN(EventLogWriter);
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_EVENT_LOG_WRITER_H_
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc b/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc
deleted file mode 100644
index 289b8e2..0000000
--- a/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc
+++ /dev/null
@@ -1,265 +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 <memory>
-
-#include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.h"
-#include "webrtc/test/gtest.h"
-
-namespace webrtc {
-
-namespace {
-
-constexpr int kMinBitrateChangeBps = 5000;
-constexpr float kMinPacketLossChangeFraction = 0.5;
-constexpr float kMinBitrateChangeFraction = 0.25;
-
-constexpr int kHighBitrateBps = 70000;
-constexpr int kLowBitrateBps = 10000;
-constexpr int kFrameLengthMs = 60;
-constexpr bool kEnableFec = true;
-constexpr bool kEnableDtx = true;
-constexpr float kPacketLossFraction = 0.05f;
-constexpr size_t kNumChannels = 1;
-
-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 EventLogWriterStates {
- std::unique_ptr<EventLogWriter> event_log_writer;
- std::unique_ptr<testing::StrictMock<MockRtcEventLog>> event_log;
- AudioNetworkAdaptor::EncoderRuntimeConfig runtime_config;
-};
-
-EventLogWriterStates CreateEventLogWriter() {
- EventLogWriterStates state;
- state.event_log.reset(new testing::StrictMock<MockRtcEventLog>());
- state.event_log_writer.reset(new EventLogWriter(
- state.event_log.get(), kMinBitrateChangeBps, kMinBitrateChangeFraction,
- kMinPacketLossChangeFraction));
- state.runtime_config.bitrate_bps = rtc::Optional<int>(kHighBitrateBps);
- state.runtime_config.frame_length_ms = rtc::Optional<int>(kFrameLengthMs);
- state.runtime_config.uplink_packet_loss_fraction =
- rtc::Optional<float>(kPacketLossFraction);
- state.runtime_config.enable_fec = rtc::Optional<bool>(kEnableFec);
- state.runtime_config.enable_dtx = rtc::Optional<bool>(kEnableDtx);
- state.runtime_config.num_channels = rtc::Optional<size_t>(kNumChannels);
- return state;
-}
-} // namespace
-
-TEST(EventLogWriterTest, FirstConfigIsLogged) {
- auto state = CreateEventLogWriter();
- EXPECT_CALL(
- *state.event_log,
- LogAudioNetworkAdaptation(EncoderRuntimeConfigIs(state.runtime_config)))
- .Times(1);
- state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
-}
-
-TEST(EventLogWriterTest, SameConfigIsNotLogged) {
- auto state = CreateEventLogWriter();
- EXPECT_CALL(
- *state.event_log,
- LogAudioNetworkAdaptation(EncoderRuntimeConfigIs(state.runtime_config)))
- .Times(1);
- state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
- state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
-}
-
-TEST(EventLogWriterTest, LogFecStateChange) {
- auto state = CreateEventLogWriter();
- EXPECT_CALL(
- *state.event_log,
- LogAudioNetworkAdaptation(EncoderRuntimeConfigIs(state.runtime_config)))
- .Times(1);
- state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
-
- state.runtime_config.enable_fec = rtc::Optional<bool>(!kEnableFec);
- EXPECT_CALL(
- *state.event_log,
- LogAudioNetworkAdaptation(EncoderRuntimeConfigIs(state.runtime_config)))
- .Times(1);
- state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
-}
-
-TEST(EventLogWriterTest, LogDtxStateChange) {
- auto state = CreateEventLogWriter();
- EXPECT_CALL(
- *state.event_log,
- LogAudioNetworkAdaptation(EncoderRuntimeConfigIs(state.runtime_config)))
- .Times(1);
- state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
-
- state.runtime_config.enable_dtx = rtc::Optional<bool>(!kEnableDtx);
- EXPECT_CALL(
- *state.event_log,
- LogAudioNetworkAdaptation(EncoderRuntimeConfigIs(state.runtime_config)))
- .Times(1);
- state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
-}
-
-TEST(EventLogWriterTest, LogChannelChange) {
- auto state = CreateEventLogWriter();
- EXPECT_CALL(
- *state.event_log,
- LogAudioNetworkAdaptation(EncoderRuntimeConfigIs(state.runtime_config)))
- .Times(1);
- state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
-
- state.runtime_config.num_channels = rtc::Optional<size_t>(kNumChannels + 1);
- EXPECT_CALL(
- *state.event_log,
- LogAudioNetworkAdaptation(EncoderRuntimeConfigIs(state.runtime_config)))
- .Times(1);
- state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
-}
-
-TEST(EventLogWriterTest, LogFrameLengthChange) {
- auto state = CreateEventLogWriter();
- EXPECT_CALL(
- *state.event_log,
- LogAudioNetworkAdaptation(EncoderRuntimeConfigIs(state.runtime_config)))
- .Times(1);
- state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
-
- state.runtime_config.frame_length_ms = rtc::Optional<int>(20);
- EXPECT_CALL(
- *state.event_log,
- LogAudioNetworkAdaptation(EncoderRuntimeConfigIs(state.runtime_config)))
- .Times(1);
- state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
-}
-
-TEST(EventLogWriterTest, DoNotLogSmallBitrateChange) {
- auto state = CreateEventLogWriter();
- EXPECT_CALL(
- *state.event_log,
- LogAudioNetworkAdaptation(EncoderRuntimeConfigIs(state.runtime_config)))
- .Times(1);
- state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
- state.runtime_config.bitrate_bps =
- rtc::Optional<int>(kHighBitrateBps + kMinBitrateChangeBps - 1);
- state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
-}
-
-TEST(EventLogWriterTest, LogLargeBitrateChange) {
- auto state = CreateEventLogWriter();
- EXPECT_CALL(
- *state.event_log,
- LogAudioNetworkAdaptation(EncoderRuntimeConfigIs(state.runtime_config)))
- .Times(1);
- state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
- // At high bitrate, the min fraction rule requires a larger change than the
- // min change rule. We make sure that the min change rule applies.
- RTC_DCHECK_GT(kHighBitrateBps * kMinBitrateChangeFraction,
- kMinBitrateChangeBps);
- state.runtime_config.bitrate_bps =
- rtc::Optional<int>(kHighBitrateBps + kMinBitrateChangeBps);
- EXPECT_CALL(
- *state.event_log,
- LogAudioNetworkAdaptation(EncoderRuntimeConfigIs(state.runtime_config)))
- .Times(1);
- state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
-}
-
-TEST(EventLogWriterTest, LogMinBitrateChangeFractionOnLowBitrateChange) {
- auto state = CreateEventLogWriter();
- state.runtime_config.bitrate_bps = rtc::Optional<int>(kLowBitrateBps);
- EXPECT_CALL(
- *state.event_log,
- LogAudioNetworkAdaptation(EncoderRuntimeConfigIs(state.runtime_config)))
- .Times(1);
- state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
- // At high bitrate, the min change rule requires a larger change than the min
- // fraction rule. We make sure that the min fraction rule applies.
- state.runtime_config.bitrate_bps = rtc::Optional<int>(
- kLowBitrateBps + kLowBitrateBps * kMinBitrateChangeFraction);
- EXPECT_CALL(
- *state.event_log,
- LogAudioNetworkAdaptation(EncoderRuntimeConfigIs(state.runtime_config)))
- .Times(1);
- state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
-}
-
-TEST(EventLogWriterTest, DoNotLogSmallPacketLossFractionChange) {
- auto state = CreateEventLogWriter();
- EXPECT_CALL(
- *state.event_log,
- LogAudioNetworkAdaptation(EncoderRuntimeConfigIs(state.runtime_config)))
- .Times(1);
- state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
- state.runtime_config.uplink_packet_loss_fraction = rtc::Optional<float>(
- kPacketLossFraction + kMinPacketLossChangeFraction * kPacketLossFraction -
- 0.001f);
- state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
-}
-
-TEST(EventLogWriterTest, LogLargePacketLossFractionChange) {
- auto state = CreateEventLogWriter();
- EXPECT_CALL(
- *state.event_log,
- LogAudioNetworkAdaptation(EncoderRuntimeConfigIs(state.runtime_config)))
- .Times(1);
- state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
- state.runtime_config.uplink_packet_loss_fraction = rtc::Optional<float>(
- kPacketLossFraction + kMinPacketLossChangeFraction * kPacketLossFraction);
- EXPECT_CALL(
- *state.event_log,
- LogAudioNetworkAdaptation(EncoderRuntimeConfigIs(state.runtime_config)))
- .Times(1);
- state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
-}
-
-TEST(EventLogWriterTest, LogJustOnceOnMultipleChanges) {
- auto state = CreateEventLogWriter();
- EXPECT_CALL(
- *state.event_log,
- LogAudioNetworkAdaptation(EncoderRuntimeConfigIs(state.runtime_config)))
- .Times(1);
- state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
- state.runtime_config.uplink_packet_loss_fraction = rtc::Optional<float>(
- kPacketLossFraction + kMinPacketLossChangeFraction * kPacketLossFraction);
- state.runtime_config.frame_length_ms = rtc::Optional<int>(20);
- EXPECT_CALL(
- *state.event_log,
- LogAudioNetworkAdaptation(EncoderRuntimeConfigIs(state.runtime_config)))
- .Times(1);
- state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
-}
-
-TEST(EventLogWriterTest, LogAfterGradualChange) {
- auto state = CreateEventLogWriter();
- EXPECT_CALL(
- *state.event_log,
- LogAudioNetworkAdaptation(EncoderRuntimeConfigIs(state.runtime_config)))
- .Times(1);
- state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
- state.runtime_config.bitrate_bps =
- rtc::Optional<int>(kHighBitrateBps + kMinBitrateChangeBps);
- EXPECT_CALL(
- *state.event_log,
- LogAudioNetworkAdaptation(EncoderRuntimeConfigIs(state.runtime_config)))
- .Times(1);
- for (int bitrate_bps = kHighBitrateBps;
- bitrate_bps <= kHighBitrateBps + kMinBitrateChangeBps; bitrate_bps++) {
- state.runtime_config.bitrate_bps = rtc::Optional<int>(bitrate_bps);
- state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
- }
-}
-} // namespace webrtc
diff --git a/webrtc/tools/BUILD.gn b/webrtc/tools/BUILD.gn
index 417e365..266f669 100644
--- a/webrtc/tools/BUILD.gn
+++ b/webrtc/tools/BUILD.gn
@@ -203,7 +203,6 @@
"../call:call_interfaces",
"../logging:rtc_event_log_impl",
"../logging:rtc_event_log_parser",
- "../modules/audio_coding:ana_debug_dump_proto",
"../modules/congestion_controller",
"../modules/rtp_rtcp",
"../system_wrappers:system_wrappers_default",
diff --git a/webrtc/tools/event_log_visualizer/analyzer.cc b/webrtc/tools/event_log_visualizer/analyzer.cc
index 3fd2c25..89e516a 100644
--- a/webrtc/tools/event_log_visualizer/analyzer.cc
+++ b/webrtc/tools/event_log_visualizer/analyzer.cc
@@ -423,9 +423,6 @@
bwe_loss_updates_.push_back(bwe_update);
break;
}
- case ParsedRtcEventLog::AUDIO_NETWORK_ADAPTATION_EVENT: {
- break;
- }
case ParsedRtcEventLog::BWE_PACKET_DELAY_EVENT: {
break;
}
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 1d39645..68f2e2d 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -140,14 +140,6 @@
}
}
- void LogAudioNetworkAdaptation(
- const AudioNetworkAdaptor::EncoderRuntimeConfig& config) override {
- rtc::CritScope lock(&crit_);
- if (event_log_) {
- event_log_->LogAudioNetworkAdaptation(config);
- }
- }
-
void SetEventLog(RtcEventLog* event_log) {
rtc::CritScope lock(&crit_);
event_log_ = event_log;