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

webrtc/modules/audio_coding/codecs/audio_encoder.cc

Index: webrtc/modules/audio_coding/codecs/audio_encoder.cc
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.cc b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
index 956c4e086f0dbf244cb348ad93083d19ad252411..560126b351539547403b4e9cc24e4470aff9f292 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.cc
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
@@ -68,6 +68,8 @@ 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;
 }
@@ -87,7 +89,4 @@ void AudioEncoder::OnReceivedTargetAudioBitrate(int 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