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

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

 /*
  *  Copyright (c) 2014 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 359 matching lines @@
   int packet_input_time_ms =
       rtp_generator_->GetRtpHeader(kPayloadType, in_size_samples_,
                                    &rtp_header_);
   Log() << "Packet of size "
         << payload_size_bytes_
         << " bytes, for frame at "
         << packet_input_time_ms
         << " ms ";
   if (payload_size_bytes_ > 0) {
     if (!PacketLost()) {
-      int ret = neteq_->InsertPacket(rtp_header_, &payload_[0],
-                                     payload_size_bytes_,
-                                     packet_input_time_ms * in_sampling_khz_);
+      int ret = neteq_->InsertPacket(
+          rtp_header_,
+          rtc::ArrayView<const uint8_t>(payload_.get(), payload_size_bytes_),
+          packet_input_time_ms * in_sampling_khz_);
       if (ret != NetEq::kOK)
         return -1;
       Log() << "was sent.";
     } else {
       Log() << "was lost.";
     }
   }
   Log() << std::endl;
   return packet_input_time_ms;
 }
@@ skipping 33 matching lines @@
     }
   }
   Log() << "Average bit rate was "
         << 8.0f * total_payload_size_bytes_ / FLAGS_runtime_ms
         << " kbps"
         << std::endl;
 }
 
 }  // namespace test
 }  // namespace webrtc