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

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

 /*
- *  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
  *  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_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 {
 
 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.
-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_