AcmReceiver::InsertPacket and NetEq::InsertPacket: Take ArrayView arguments

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 529 matching lines @@
         payload_len = ReplacePayload(replacement_audio_file.get(),
                                      &replacement_audio,
                                      &payload,
                                      &payload_mem_size_bytes,
                                      &input_frame_size_timestamps,
                                      &rtp_header,
                                      next_packet.get());
         payload_ptr = payload.get();
       }
       int error = neteq->InsertPacket(
-          rtp_header, payload_ptr, payload_len,
+          rtp_header, rtc::ArrayView<const uint8_t>(payload_ptr, payload_len),
           static_cast<uint32_t>(packet->time_ms() * sample_rate_hz / 1000));
       if (error != NetEq::kOK) {
         if (neteq->LastError() == NetEq::kUnknownRtpPayloadType) {
           std::cerr << "RTP Payload type "
                     << static_cast<int>(rtp_header.header.payloadType)
                     << " is unknown." << std::endl;
           std::cerr << "Use --codec_map to view default mapping." << std::endl;
           std::cerr << "Use --helpshort for information on how to make custom "
                        "mappings." << std::endl;
         } else {
@@ skipping 70 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;
 }