diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index 8f87576e8f..b2839be12a 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -933,7 +933,6 @@ rtc_static_library("audio_network_adaptor") {
       ":ana_config_proto",
       ":ana_debug_dump_proto",
     ]
-    defines = [ "WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP" ]
   }
 
   if (!build_with_chromium && is_clang) {
@@ -2178,10 +2177,7 @@ if (rtc_include_tests) {
     defines = audio_coding_defines
 
     if (rtc_enable_protobuf) {
-      defines += [
-        "WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP",
-        "WEBRTC_NETEQ_UNITTEST_BITEXACT",
-      ]
+      defines += [ "WEBRTC_NETEQ_UNITTEST_BITEXACT" ]
       deps += [
         ":ana_config_proto",
         ":neteq_unittest_proto",
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc b/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc
index c445ebe3b2..8cb142f2c9 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc
@@ -23,7 +23,7 @@
 #include "webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/util/threshold_curve.h"
 
-#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#if WEBRTC_ENABLE_PROTOBUF
 RTC_PUSH_IGNORING_WUNDEF()
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h"
@@ -37,7 +37,7 @@ namespace webrtc {
 
 namespace {
 
-#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#if WEBRTC_ENABLE_PROTOBUF
 
 std::unique_ptr<FecControllerPlrBased> CreateFecControllerPlrBased(
     const audio_network_adaptor::config::FecController& config,
@@ -180,7 +180,7 @@ std::unique_ptr<BitrateController> CreateBitrateController(
   return std::unique_ptr<BitrateController>(new BitrateController(
       BitrateController::Config(initial_bitrate_bps, initial_frame_length_ms)));
 }
-#endif  // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#endif  // WEBRTC_ENABLE_PROTOBUF
 
 }  // namespace
 
@@ -201,7 +201,7 @@ std::unique_ptr<ControllerManager> ControllerManagerImpl::Create(
     int initial_bitrate_bps,
     bool initial_fec_enabled,
     bool initial_dtx_enabled) {
-#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#if WEBRTC_ENABLE_PROTOBUF
   audio_network_adaptor::config::ControllerManager controller_manager_config;
   controller_manager_config.ParseFromString(config_string);
 
@@ -270,7 +270,7 @@ std::unique_ptr<ControllerManager> ControllerManagerImpl::Create(
 #else
   RTC_NOTREACHED();
   return nullptr;
-#endif  // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#endif  // WEBRTC_ENABLE_PROTOBUF
 }
 
 ControllerManagerImpl::ControllerManagerImpl(const Config& config)
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc b/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
index 52f15835d8..a2deb7bfbb 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
@@ -17,7 +17,7 @@
 #include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller.h"
 #include "webrtc/test/gtest.h"
 
-#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#if WEBRTC_ENABLE_PROTOBUF
 RTC_PUSH_IGNORING_WUNDEF()
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h"
@@ -203,7 +203,7 @@ TEST(ControllerManagerTest, DoNotReorderIfNetworkMetricsChangeTooSmall) {
       {kNumControllers - 2, kNumControllers - 1, 0, 1});
 }
 
-#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#if WEBRTC_ENABLE_PROTOBUF
 
 namespace {
 
@@ -439,6 +439,6 @@ TEST(ControllerManagerTest, CreateFromConfigStringAndCheckReordering) {
                             ControllerType::CHANNEL, ControllerType::DTX,
                             ControllerType::BIT_RATE});
 }
-#endif  // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#endif  // WEBRTC_ENABLE_PROTOBUF
 
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
index 8b485e4a07..fdedf6cd9d 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
@@ -14,7 +14,7 @@
 #include "webrtc/base/ignore_wundef.h"
 #include "webrtc/base/protobuf_utils.h"
 
-#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#if WEBRTC_ENABLE_PROTOBUF
 RTC_PUSH_IGNORING_WUNDEF()
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.pb.h"
@@ -26,7 +26,7 @@ RTC_POP_IGNORING_WUNDEF()
 
 namespace webrtc {
 
-#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#if WEBRTC_ENABLE_PROTOBUF
 namespace {
 
 using audio_network_adaptor::debug_dump::Event;
@@ -43,7 +43,7 @@ void DumpEventToFile(const Event& event, FileWrapper* dump_file) {
 }
 
 }  // namespace
-#endif  // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#endif  // WEBRTC_ENABLE_PROTOBUF
 
 class DebugDumpWriterImpl final : public DebugDumpWriter {
  public:
@@ -62,17 +62,18 @@ class DebugDumpWriterImpl final : public DebugDumpWriter {
 
 DebugDumpWriterImpl::DebugDumpWriterImpl(FILE* file_handle)
     : dump_file_(FileWrapper::Create()) {
-#ifndef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
-  RTC_NOTREACHED();
-#endif
+#if WEBRTC_ENABLE_PROTOBUF
   dump_file_->OpenFromFileHandle(file_handle);
   RTC_CHECK(dump_file_->is_open());
+#else
+  RTC_NOTREACHED();
+#endif
 }
 
 void DebugDumpWriterImpl::DumpNetworkMetrics(
     const Controller::NetworkMetrics& metrics,
     int64_t timestamp) {
-#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#if WEBRTC_ENABLE_PROTOBUF
   Event event;
   event.set_timestamp(timestamp);
   event.set_type(Event::NETWORK_METRICS);
@@ -100,13 +101,13 @@ void DebugDumpWriterImpl::DumpNetworkMetrics(
   }
 
   DumpEventToFile(event, dump_file_.get());
-#endif  // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#endif  // WEBRTC_ENABLE_PROTOBUF
 }
 
 void DebugDumpWriterImpl::DumpEncoderRuntimeConfig(
     const AudioEncoderRuntimeConfig& config,
     int64_t timestamp) {
-#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#if WEBRTC_ENABLE_PROTOBUF
   Event event;
   event.set_timestamp(timestamp);
   event.set_type(Event::ENCODER_RUNTIME_CONFIG);
@@ -133,7 +134,7 @@ void DebugDumpWriterImpl::DumpEncoderRuntimeConfig(
     dump_config->set_num_channels(*config.num_channels);
 
   DumpEventToFile(event, dump_file_.get());
-#endif  // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+#endif  // WEBRTC_ENABLE_PROTOBUF
 }
 
 std::unique_ptr<DebugDumpWriter> DebugDumpWriter::Create(FILE* file_handle) {