diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index 898723b8ca..d415098e98 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -881,9 +881,18 @@ if (rtc_enable_protobuf) {
   }
 }
 
+rtc_static_library("audio_network_adaptor_config") {
+  sources = [
+    "audio_network_adaptor/audio_network_adaptor_config.cc",
+    "audio_network_adaptor/include/audio_network_adaptor_config.h",
+  ]
+  deps = [
+    "../../api:optional",
+  ]
+}
+
 rtc_static_library("audio_network_adaptor") {
   sources = [
-    "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",
@@ -910,6 +919,10 @@ rtc_static_library("audio_network_adaptor") {
     "audio_network_adaptor/util/threshold_curve.h",
   ]
 
+  public_deps = [
+    ":audio_network_adaptor_config",
+  ]
+
   deps = [
     "../..:webrtc_common",
     "../../api:optional",
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor.cc b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_config.cc
similarity index 95%
rename from webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor.cc
rename to webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_config.cc
index ce1e2500ff..c312266d8a 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_config.cc
@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
+#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
 
 namespace webrtc {
 
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h b/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
index 0097d70bd8..a91b33b34f 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
@@ -13,27 +13,10 @@
 
 #include "webrtc/api/audio_codecs/audio_encoder.h"
 #include "webrtc/api/optional.h"
+#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
 
 namespace webrtc {
 
-struct AudioEncoderRuntimeConfig {
-  AudioEncoderRuntimeConfig();
-  AudioEncoderRuntimeConfig(const AudioEncoderRuntimeConfig& other);
-  ~AudioEncoderRuntimeConfig();
-  rtc::Optional<int> bitrate_bps;
-  rtc::Optional<int> frame_length_ms;
-  // Note: This is what we tell the encoder. It doesn't have to reflect
-  // the actual NetworkMetrics; it's subject to our decision.
-  rtc::Optional<float> uplink_packet_loss_fraction;
-  rtc::Optional<bool> enable_fec;
-  rtc::Optional<bool> enable_dtx;
-
-  // Some encoders can encode fewer channels than the actual input to make
-  // better use of the bandwidth. |num_channels| sets the number of channels
-  // to encode.
-  rtc::Optional<size_t> num_channels;
-};
-
 // An AudioNetworkAdaptor optimizes the audio experience by suggesting a
 // suitable runtime configuration (bit rate, frame length, FEC, etc.) to the
 // encoder based on network metrics.
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h b/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h
new file mode 100644
index 0000000000..5fc68e6522
--- /dev/null
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2017 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_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_
+#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_
+
+#include "webrtc/api/optional.h"
+
+namespace webrtc {
+
+struct AudioEncoderRuntimeConfig {
+  AudioEncoderRuntimeConfig();
+  AudioEncoderRuntimeConfig(const AudioEncoderRuntimeConfig& other);
+  ~AudioEncoderRuntimeConfig();
+  rtc::Optional<int> bitrate_bps;
+  rtc::Optional<int> frame_length_ms;
+  // Note: This is what we tell the encoder. It doesn't have to reflect
+  // the actual NetworkMetrics; it's subject to our decision.
+  rtc::Optional<float> uplink_packet_loss_fraction;
+  rtc::Optional<bool> enable_fec;
+  rtc::Optional<bool> enable_dtx;
+
+  // Some encoders can encode fewer channels than the actual input to make
+  // better use of the bandwidth. |num_channels| sets the number of channels
+  // to encode.
+  rtc::Optional<size_t> num_channels;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_