Remove the type parameter to NetEq::GetAudio

webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc

 /*
  *  Copyright (c) 2013 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 587 matching lines @@
         // Set next input time to the maximum value of int64_t to prevent the
         // time_now_ms from becoming stuck at the final value.
         next_input_time_ms = std::numeric_limits<int64_t>::max();
         packet_available = false;
       }
     }
 
     // Check if it is time to get output audio.
     while (time_now_ms >= next_output_time_ms && output_event_available) {
       webrtc::AudioFrame out_frame;
-      int error = neteq->GetAudio(&out_frame, NULL);
+      int error = neteq->GetAudio(&out_frame);
       if (error != NetEq::kOK) {
         std::cerr << "GetAudio returned error code " <<
             neteq->LastError() << std::endl;
       } else {
         sample_rate_hz = out_frame.sample_rate_hz_;
       }
 
       // Write to file.
       // TODO(hlundin): Make writing to file optional.
       if (!output->WriteArray(out_frame.data_, out_frame.samples_per_channel_ *
@@ skipping 14 matching lines @@
     }
   }
   printf("Simulation done\n");
   printf("Produced %i ms of audio\n",
          static_cast<int>(time_now_ms - start_time_ms));
 
   delete neteq;
   webrtc::Trace::ReturnTrace();
   return 0;
 }