diff --git a/webrtc/logging/BUILD.gn b/webrtc/logging/BUILD.gn
index 8bce39a561..2daaef4ffc 100644
--- a/webrtc/logging/BUILD.gn
+++ b/webrtc/logging/BUILD.gn
@@ -42,7 +42,6 @@ rtc_static_library("rtc_event_log_impl") {
     ":rtc_event_log_api",
     "..:webrtc_common",
     "../call:call_interfaces",
-    "../modules/audio_coding:audio_network_adaptor",
     "../modules/rtp_rtcp",
   ]
 
@@ -85,8 +84,7 @@ if (rtc_enable_protobuf) {
     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 e340150be7..39d2020bee 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 4db33a13ec..315967b8c0 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 @@ class MockRtcEventLog : public RtcEventLog {
                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 65ee7d8881..21ca5e127d 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 @@ class RtcEventLogImpl final : public RtcEventLog {
   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 @@ void RtcEventLogImpl::LogBwePacketLossEvent(int32_t bitrate,
   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 5d221d4ef8..ccf60944e2 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 @@ class RtcEventLog {
                                      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 @@ class RtcEventLogNullImpl final : public RtcEventLog {
   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 e807722979..a6d1695796 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 @@ message Event {
     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 @@ message Event {
 
   // 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 @@ message AudioSendConfig {
   // 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 3b808b2e43..ce55a4fb39 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 @@ ParsedRtcEventLog::EventType GetRuntimeEventType(
       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::GetBwePacketLossEvent(size_t index,
   }
 }
 
-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 8472668d75..2d66b9054b 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 @@ class ParsedRtcEventLog {
     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 @@ class ParsedRtcEventLog {
                              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 f0ee973a13..1540466785 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 GenerateAudioSendConfig(uint32_t extensions_bitvector,
   }
 }
 
-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 @@ class VideoSendConfigReadWriteTest : public ConfigReadWriteTest {
   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 @@ TEST(RtcEventLogTest, LogVideoSendConfig) {
   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 19ca8aa15c..fb60dbcc05 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 @@ void RtcEventLogTestHelper::VerifyBweLossEvent(
   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 2f4e17703b..3f89e304b2 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 @@ class RtcEventLogTestHelper {
                                  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 2b941a813d..f3b2ac9aaa 100644
--- a/webrtc/modules/BUILD.gn
+++ b/webrtc/modules/BUILD.gn
@@ -258,7 +258,6 @@ if (rtc_include_tests) {
       "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 a8799be7c6..c38cd4761f 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -887,8 +887,7 @@ if (rtc_enable_protobuf) {
     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 @@ rtc_static_library("audio_network_adaptor") {
     "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 @@ rtc_static_library("audio_network_adaptor") {
     "../..:webrtc_common",
     "../../base:rtc_base_approved",
     "../../common_audio",
-    "../../logging:rtc_event_log_api",
     "../../system_wrappers",
   ]
 
@@ -939,11 +935,6 @@ rtc_static_library("audio_network_adaptor") {
     ]
     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 6ec92b00fc..3b767d9b59 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 @@ AudioNetworkAdaptorImpl::AudioNetworkAdaptorImpl(
     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 @@ AudioNetworkAdaptorImpl::GetEncoderRuntimeConfig() {
        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 9c47125da4..801a9ccc59 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 @@ class AudioNetworkAdaptorImpl final : public AudioNetworkAdaptor {
 
   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 9b608dce47..b26d2f5db8 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 @@ struct AudioNetworkAdaptorStates {
   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 @@ AudioNetworkAdaptorStates CreateAudioNetworkAdaptor() {
       .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 @@ AudioNetworkAdaptorStates CreateAudioNetworkAdaptor() {
 
   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 @@ TEST(AudioNetworkAdaptorImplTest,
   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 619a2473d9..0000000000
--- 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 740da8c31a..0000000000
--- 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 289b8e2c7a..0000000000
--- 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 417e365489..266f6690c5 100644
--- a/webrtc/tools/BUILD.gn
+++ b/webrtc/tools/BUILD.gn
@@ -203,7 +203,6 @@ if (rtc_enable_protobuf) {
       "../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 3fd2c25fe8..89e516a950 100644
--- a/webrtc/tools/event_log_visualizer/analyzer.cc
+++ b/webrtc/tools/event_log_visualizer/analyzer.cc
@@ -423,9 +423,6 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log)
         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 1d39645d47..68f2e2d95c 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -140,14 +140,6 @@ class RtcEventLogProxy final : public webrtc::RtcEventLog {
     }
   }
 
-  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;