Break the ANA build-target into ANA and ANA-config

webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h

Index: webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h
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_config.h
similarity index 54%
copy from webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
copy to webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h
index 0097d70bd8e75cd56dcc17c849f4a012d57b7297..5fc68e65222897e871d06f826ec604b95d387d9a 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_config.h
@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *  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
@@ -8,10 +8,9 @@
  *  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_H_
-#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
+#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/audio_codecs/audio_encoder.h"
 #include "webrtc/api/optional.h"
 
 namespace webrtc {
@@ -34,36 +33,6 @@ struct AudioEncoderRuntimeConfig {
   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.
-class AudioNetworkAdaptor {
- public:
-  virtual ~AudioNetworkAdaptor() = default;
-
-  virtual void SetUplinkBandwidth(int uplink_bandwidth_bps) = 0;
-
-  virtual void SetUplinkPacketLossFraction(
-      float uplink_packet_loss_fraction) = 0;
-
-  virtual void SetUplinkRecoverablePacketLossFraction(
-      float uplink_recoverable_packet_loss_fraction) = 0;
-
-  virtual void SetRtt(int rtt_ms) = 0;
-
-  virtual void SetTargetAudioBitrate(int target_audio_bitrate_bps) = 0;
-
-  virtual void SetOverhead(size_t overhead_bytes_per_packet) = 0;
-
-  virtual AudioEncoderRuntimeConfig GetEncoderRuntimeConfig() = 0;
-
-  virtual void StartDebugDump(FILE* file_handle) = 0;
-
-  virtual void StopDebugDump() = 0;
-
-  virtual ANAStats GetStats() const = 0;
-};
-
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
+#endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_