Simplifying audio network adaptor by moving receiver frame length range to ctor.

webrtc/modules/audio_coding/codecs/audio_encoder.cc

 /*
  *  Copyright (c) 2014 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.
  */
 
@@ skipping 50 matching lines @@
 void AudioEncoder::SetMaxPlaybackRate(int frequency_hz) {}
 
 void AudioEncoder::SetProjectedPacketLossRate(double fraction) {}
 
 void AudioEncoder::SetTargetBitrate(int target_bps) {}
 
 rtc::ArrayView<std::unique_ptr<AudioEncoder>>
 AudioEncoder::ReclaimContainedEncoders() { return nullptr; }
 
 bool AudioEncoder::EnableAudioNetworkAdaptor(const std::string& config_string,
+                                             int min_receiver_frame_length_ms,
+                                             int max_receiver_frame_length_ms,
                                              const Clock* clock) {
   return false;
 }
 
 void AudioEncoder::DisableAudioNetworkAdaptor() {}
 
 void AudioEncoder::OnReceivedUplinkBandwidth(int uplink_bandwidth_bps) {}
 
 void AudioEncoder::OnReceivedUplinkPacketLossFraction(
     float uplink_packet_loss_fraction) {
   SetProjectedPacketLossRate(uplink_packet_loss_fraction);
 }
 
 void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) {
   SetTargetBitrate(target_audio_bitrate_bps);
 }
 
 void AudioEncoder::OnReceivedRtt(int rtt_ms) {}
 
-void AudioEncoder::SetReceiverFrameLengthRange(int min_frame_length_ms,
-                                               int max_frame_length_ms) {}
-
 }  // namespace webrtc