Setting up an RTP input fuzzer for NetEq

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

 /*
  *  Copyright (c) 2016 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 47 matching lines @@
   // returned one is pre-fetched in the NetEqInput object, such that future
   // calls to NextPacketTime() or NextHeader() will return information from that
   // packet.
   virtual std::unique_ptr<PacketData> PopPacket() = 0;
 
   // Move to the next output event. This will make NextOutputEventTime() return
   // a new value (potentially the same if several output events share the same
   // 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
   // delivered next by PopPacket().
   virtual rtc::Optional<RTPHeader> NextHeader() const = 0;
 };
 
 }  // namespace test
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_