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

webrtc/modules/audio_coding/audio_network_adaptor/controller.h

Index: webrtc/modules/audio_coding/audio_network_adaptor/controller.h
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/controller.h b/webrtc/modules/audio_coding/audio_network_adaptor/controller.h
index f27a391ad543b55bb19e0af2b23e06ada503ab0e..07806aa000ca85e3ad723d15ab689a3edb7867a1 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/controller.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/controller.h
@@ -27,25 +27,11 @@ class Controller {
     rtc::Optional<int> rtt_ms;
   };
 
-  struct Constraints {
-    Constraints();
-    ~Constraints();
-    struct FrameLengthRange {
-      FrameLengthRange(int min_frame_length_ms, int max_frame_length_ms);
-      ~FrameLengthRange();
-      int min_frame_length_ms;
-      int max_frame_length_ms;
-    };
-    rtc::Optional<FrameLengthRange> receiver_frame_length_range;
-  };
-
   virtual ~Controller() = default;
 
   virtual void MakeDecision(
       const NetworkMetrics& metrics,
       AudioNetworkAdaptor::EncoderRuntimeConfig* config) = 0;
-
-  virtual void SetConstraints(const Constraints& constraints);
 };
 
 }  // namespace webrtc