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

webrtc/modules/audio_coding/codecs/audio_encoder.h

Index: webrtc/modules/audio_coding/codecs/audio_encoder.h
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h
index 19dc3327578ec3c1853da96599052612f82b3e77..9388be1bc13a687cdb819b38a6a4a9071ec96e0c 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.h
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h
@@ -166,6 +166,8 @@ class AudioEncoder {
 
   // Enables audio network adaptor. Returns true if successful.
   virtual bool EnableAudioNetworkAdaptor(const std::string& config_string,
+                                         int min_receiver_frame_length_ms,
+                                         int max_receiver_frame_length_ms,
                                          const Clock* clock);
 
   // Disables audio network adaptor.
@@ -184,11 +186,6 @@ class AudioEncoder {
   // Provides RTT to this encoder to allow it to adapt.
   virtual void OnReceivedRtt(int rtt_ms);
 
-  // To allow encoder to adapt its frame length, it must be provided the frame
-  // length range that receives can accept.
-  virtual void SetReceiverFrameLengthRange(int min_frame_length_ms,
-                                           int max_frame_length_ms);
-
  protected:
   // Subclasses implement this to perform the actual encoding. Called by
   // Encode().