diff --git a/webrtc/modules/BUILD.gn b/webrtc/modules/BUILD.gn
index 09983c2..38aebcd 100644
--- a/webrtc/modules/BUILD.gn
+++ b/webrtc/modules/BUILD.gn
@@ -233,31 +233,6 @@
     }
   }
 
-  rtc_source_set("audio_network_adaptor_unittests") {
-    # Put sources for unittests of audio network adaptor in a separate
-    # rtc_source_set to solve name collision on bitrate_controller_unittest.cc.
-    testonly = true
-    sources = [
-      "audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc",
-      "audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc",
-      "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/mock/mock_controller.h",
-      "audio_coding/audio_network_adaptor/mock/mock_controller_manager.h",
-      "audio_coding/audio_network_adaptor/smoothing_filter_unittest.cc",
-    ]
-    deps = [
-      "audio_coding:audio_network_adaptor",
-      "//testing/gmock",
-      "//testing/gtest",
-    ]
-    if (is_clang) {
-      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
-    }
-  }
-
   rtc_test("modules_unittests") {
     testonly = true
 
@@ -272,6 +247,13 @@
       "audio_coding/acm2/codec_manager_unittest.cc",
       "audio_coding/acm2/initial_delay_manager_unittest.cc",
       "audio_coding/acm2/rent_a_codec_unittest.cc",
+      "audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc",
+      "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/mock/mock_controller.h",
+      "audio_coding/audio_network_adaptor/mock/mock_controller_manager.h",
+      "audio_coding/audio_network_adaptor/smoothing_filter_unittest.cc",
       "audio_coding/codecs/audio_decoder_factory_unittest.cc",
       "audio_coding/codecs/cng/audio_encoder_cng_unittest.cc",
       "audio_coding/codecs/cng/cng_unittest.cc",
@@ -615,7 +597,6 @@
     }
 
     deps += [
-      ":audio_network_adaptor_unittests",
       "..:webrtc_common",
       "../base:rtc_base",  # TODO(kjellander): Cleanup in bugs.webrtc.org/3806.
       "../common_audio",
@@ -629,6 +610,7 @@
       "audio_coding",
       "audio_coding:acm_receive_test",
       "audio_coding:acm_send_test",
+      "audio_coding:audio_network_adaptor",
       "audio_coding:builtin_audio_decoder_factory",
       "audio_coding:cng",
       "audio_coding:isac_fix",
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index a7b50a0..09202af 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -709,8 +709,6 @@
     "audio_network_adaptor/audio_network_adaptor.cc",
     "audio_network_adaptor/audio_network_adaptor_impl.cc",
     "audio_network_adaptor/audio_network_adaptor_impl.h",
-    "audio_network_adaptor/bitrate_controller.cc",
-    "audio_network_adaptor/bitrate_controller.h",
     "audio_network_adaptor/channel_controller.cc",
     "audio_network_adaptor/channel_controller.h",
     "audio_network_adaptor/controller.cc",
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 2a591c3..d266ddb 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
@@ -14,8 +14,6 @@
         'audio_network_adaptor.cc',
         'audio_network_adaptor_impl.cc',
         'audio_network_adaptor_impl.h',
-        'bitrate_controller.h',
-        'bitrate_controller.cc',
         'channel_controller.cc',
         'channel_controller.h',
         'controller.h',
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..c17d389 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
@@ -28,7 +28,6 @@
 
 MATCHER_P(NetworkMetricsIs, metric, "") {
   return arg.uplink_bandwidth_bps == metric.uplink_bandwidth_bps &&
-         arg.target_audio_bitrate_bps == metric.target_audio_bitrate_bps &&
          arg.uplink_packet_loss_fraction == metric.uplink_packet_loss_fraction;
 }
 
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc b/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
deleted file mode 100644
index 4cc49e8..0000000
--- a/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.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 "webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
-
-#include <algorithm>
-
-#include "webrtc/base/checks.h"
-
-namespace webrtc {
-
-namespace {
-// TODO(minyue): consider passing this from a higher layer through
-// SetConstraints().
-// L2(14B) + IPv4(20B) + UDP(8B) + RTP(12B) + SRTP_AUTH(10B) = 64B = 512 bits
-constexpr int kPacketOverheadBits = 512;
-}
-
-BitrateController::Config::Config(int initial_bitrate_bps,
-                                  int initial_frame_length_ms)
-    : initial_bitrate_bps(initial_bitrate_bps),
-      initial_frame_length_ms(initial_frame_length_ms) {}
-
-BitrateController::Config::~Config() = default;
-
-BitrateController::BitrateController(const Config& config)
-    : config_(config),
-      bitrate_bps_(config_.initial_bitrate_bps),
-      overhead_rate_bps_(kPacketOverheadBits * 1000 /
-                         config_.initial_frame_length_ms) {
-  RTC_DCHECK_GT(bitrate_bps_, 0);
-  RTC_DCHECK_GT(overhead_rate_bps_, 0);
-}
-
-void BitrateController::MakeDecision(
-    const NetworkMetrics& metrics,
-    AudioNetworkAdaptor::EncoderRuntimeConfig* config) {
-  // Decision on |bitrate_bps| should not have been made.
-  RTC_DCHECK(!config->bitrate_bps);
-
-  if (metrics.target_audio_bitrate_bps) {
-    int overhead_rate =
-        config->frame_length_ms
-            ? kPacketOverheadBits * 1000 / *config->frame_length_ms
-            : overhead_rate_bps_;
-    // If |metrics.target_audio_bitrate_bps| had included overhead, we would
-    // simply do:
-    //     bitrate_bps_ = metrics.target_audio_bitrate_bps - overhead_rate;
-    // Follow https://bugs.chromium.org/p/webrtc/issues/detail?id=6315 to track
-    // progress regarding this.
-    // Now we assume that |metrics.target_audio_bitrate_bps| can handle the
-    // overhead of most recent packets.
-    bitrate_bps_ = std::max(0, *metrics.target_audio_bitrate_bps +
-                                   overhead_rate_bps_ - overhead_rate);
-    // TODO(minyue): apply a smoothing on the |overhead_rate_bps_|.
-    overhead_rate_bps_ = overhead_rate;
-  }
-  config->bitrate_bps = rtc::Optional<int>(bitrate_bps_);
-}
-
-}  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h b/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h
deleted file mode 100644
index cfc6fa8..0000000
--- a/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h
+++ /dev/null
@@ -1,42 +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_BITRATE_CONTROLLER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_BITRATE_CONTROLLER_H_
-
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/controller.h"
-
-namespace webrtc {
-
-class BitrateController final : public Controller {
- public:
-  struct Config {
-    Config(int initial_bitrate_bps, int frame_length_ms);
-    ~Config();
-    int initial_bitrate_bps;
-    int initial_frame_length_ms;
-  };
-
-  explicit BitrateController(const Config& config);
-
-  void MakeDecision(const NetworkMetrics& metrics,
-                    AudioNetworkAdaptor::EncoderRuntimeConfig* config) override;
-
- private:
-  const Config config_;
-  int bitrate_bps_;
-  int overhead_rate_bps_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(BitrateController);
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_BITRATE_CONTROLLER_H_
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc b/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
deleted file mode 100644
index 5a3a74a..0000000
--- a/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
+++ /dev/null
@@ -1,142 +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 "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
-
-namespace webrtc {
-
-namespace {
-
-// L2(14B) + IPv4(20B) + UDP(8B) + RTP(12B) + SRTP_AUTH(10B) = 64B = 512 bits
-constexpr int kPacketOverheadBits = 512;
-
-void CheckDecision(BitrateController* controller,
-                   const rtc::Optional<int>& target_audio_bitrate_bps,
-                   const rtc::Optional<int>& frame_length_ms,
-                   int expected_bitrate_bps) {
-  Controller::NetworkMetrics metrics;
-  metrics.target_audio_bitrate_bps = target_audio_bitrate_bps;
-  AudioNetworkAdaptor::EncoderRuntimeConfig config;
-  config.frame_length_ms = frame_length_ms;
-  controller->MakeDecision(metrics, &config);
-  EXPECT_EQ(rtc::Optional<int>(expected_bitrate_bps), config.bitrate_bps);
-}
-
-}  // namespace
-
-// These tests are named AnaBitrateControllerTest to distinguish from
-// BitrateControllerTest in
-// modules/bitrate_controller/bitrate_controller_unittest.cc.
-
-TEST(AnaBitrateControllerTest, OutputInitValueWhenTargetBitrateUnknown) {
-  constexpr int kInitialBitrateBps = 32000;
-  constexpr int kInitialFrameLengthMs = 20;
-  BitrateController controller(
-      BitrateController::Config(kInitialBitrateBps, kInitialFrameLengthMs));
-  CheckDecision(&controller, rtc::Optional<int>(),
-                rtc::Optional<int>(kInitialFrameLengthMs * 2),
-                kInitialBitrateBps);
-}
-
-TEST(AnaBitrateControllerTest, ChangeBitrateOnTargetBitrateChanged) {
-  constexpr int kInitialBitrateBps = 32000;
-  constexpr int kInitialFrameLengthMs = 20;
-  BitrateController controller(
-      BitrateController::Config(kInitialBitrateBps, kInitialFrameLengthMs));
-  constexpr int kTargetBitrateBps = 48000;
-  // Frame length unchanged, bitrate changes in accordance with
-  // |metrics.target_audio_bitrate_bps|
-  CheckDecision(&controller, rtc::Optional<int>(kTargetBitrateBps),
-                rtc::Optional<int>(kInitialFrameLengthMs), kTargetBitrateBps);
-}
-
-TEST(AnaBitrateControllerTest, TreatUnknownFrameLengthAsFrameLengthUnchanged) {
-  constexpr int kInitialBitrateBps = 32000;
-  constexpr int kInitialFrameLengthMs = 20;
-  BitrateController controller(
-      BitrateController::Config(kInitialBitrateBps, kInitialFrameLengthMs));
-  constexpr int kTargetBitrateBps = 48000;
-  CheckDecision(&controller, rtc::Optional<int>(kTargetBitrateBps),
-                rtc::Optional<int>(), kTargetBitrateBps);
-}
-
-TEST(AnaBitrateControllerTest, IncreaseBitrateOnFrameLengthIncreased) {
-  constexpr int kInitialBitrateBps = 32000;
-  constexpr int kInitialFrameLengthMs = 20;
-  BitrateController controller(
-      BitrateController::Config(kInitialBitrateBps, kInitialFrameLengthMs));
-  constexpr int kFrameLengthMs = 60;
-  constexpr int kPacketOverheadRateDiff =
-      kPacketOverheadBits * 1000 / kInitialFrameLengthMs -
-      kPacketOverheadBits * 1000 / kFrameLengthMs;
-  CheckDecision(&controller, rtc::Optional<int>(kInitialBitrateBps),
-                rtc::Optional<int>(kFrameLengthMs),
-                kInitialBitrateBps + kPacketOverheadRateDiff);
-}
-
-TEST(AnaBitrateControllerTest, DecreaseBitrateOnFrameLengthDecreased) {
-  constexpr int kInitialBitrateBps = 32000;
-  constexpr int kInitialFrameLengthMs = 60;
-  BitrateController controller(
-      BitrateController::Config(kInitialBitrateBps, kInitialFrameLengthMs));
-  constexpr int kFrameLengthMs = 20;
-  constexpr int kPacketOverheadRateDiff =
-      kPacketOverheadBits * 1000 / kInitialFrameLengthMs -
-      kPacketOverheadBits * 1000 / kFrameLengthMs;
-  CheckDecision(&controller, rtc::Optional<int>(kInitialBitrateBps),
-                rtc::Optional<int>(kFrameLengthMs),
-                kInitialBitrateBps + kPacketOverheadRateDiff);
-}
-
-TEST(AnaBitrateControllerTest, CheckBehaviorOnChangingCondition) {
-  constexpr int kInitialBitrateBps = 32000;
-  constexpr int kInitialFrameLengthMs = 20;
-  BitrateController controller(
-      BitrateController::Config(kInitialBitrateBps, kInitialFrameLengthMs));
-
-  int last_overhead_bitrate =
-      kPacketOverheadBits * 1000 / kInitialFrameLengthMs;
-  int current_overhead_bitrate = kPacketOverheadBits * 1000 / 20;
-  // Start from an arbitrary overall bitrate.
-  int overall_bitrate = 34567;
-  CheckDecision(
-      &controller, rtc::Optional<int>(overall_bitrate - last_overhead_bitrate),
-      rtc::Optional<int>(20), overall_bitrate - current_overhead_bitrate);
-
-  // Next: increase overall bitrate.
-  overall_bitrate += 100;
-  CheckDecision(
-      &controller, rtc::Optional<int>(overall_bitrate - last_overhead_bitrate),
-      rtc::Optional<int>(20), overall_bitrate - current_overhead_bitrate);
-
-  // Next: change frame length.
-  current_overhead_bitrate = kPacketOverheadBits * 1000 / 60;
-  CheckDecision(
-      &controller, rtc::Optional<int>(overall_bitrate - last_overhead_bitrate),
-      rtc::Optional<int>(60), overall_bitrate - current_overhead_bitrate);
-  last_overhead_bitrate = current_overhead_bitrate;
-
-  // Next: change frame length.
-  current_overhead_bitrate = kPacketOverheadBits * 1000 / 20;
-  CheckDecision(
-      &controller, rtc::Optional<int>(overall_bitrate - last_overhead_bitrate),
-      rtc::Optional<int>(20), overall_bitrate - current_overhead_bitrate);
-  last_overhead_bitrate = current_overhead_bitrate;
-
-  // Next: decrease overall bitrate and frame length.
-  overall_bitrate -= 100;
-  current_overhead_bitrate = kPacketOverheadBits * 1000 / 60;
-  CheckDecision(
-      &controller, rtc::Optional<int>(overall_bitrate - last_overhead_bitrate),
-      rtc::Optional<int>(60), overall_bitrate - current_overhead_bitrate);
-}
-
-}  // 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..f6a6079 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/controller.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/controller.h
@@ -23,7 +23,6 @@
     ~NetworkMetrics();
     rtc::Optional<int> uplink_bandwidth_bps;
     rtc::Optional<float> uplink_packet_loss_fraction;
-    rtc::Optional<int> target_audio_bitrate_bps;
   };
 
   struct Constraints {