Setting up an RTP input fuzzer for NetEq

webrtc/modules/audio_coding/neteq/tools/neteq_input.h

Index: webrtc/modules/audio_coding/neteq/tools/neteq_input.h
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_input.h b/webrtc/modules/audio_coding/neteq/tools/neteq_input.h
index 8abec6303fc70c4e01fd24dfc2cfac97bfef896c..be08a791a81e3168ad680040f30135aedca174c4 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_input.h
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_input.h
@@ -65,7 +65,9 @@ class NetEqInput {
   // time).
   virtual void AdvanceOutputEvent() = 0;
 
-  // Returns true if the source has come to an end.
+  // Returns true if the source has come to an end. An implementation must
+  // eventually return true from this method, or the test will end up in an
+  // infinite loop.
   virtual bool ended() const = 0;
 
   // Returns the RTP header for the next packet, i.e., the packet that will be