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

webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc

 /*
  *  Copyright (c) 2012 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 593 matching lines @@
 // Get current received codec.
 int AudioCodingModuleImpl::ReceiveCodec(CodecInst* current_codec) const {
   CriticalSectionScoped lock(acm_crit_sect_.get());
   return receiver_.LastAudioCodec(current_codec);
 }
 
 // Incoming packet from network parsed and ready for decode.
 int AudioCodingModuleImpl::IncomingPacket(const uint8_t* incoming_payload,
                                           const size_t payload_length,
                                           const WebRtcRTPHeader& rtp_header) {
-  return receiver_.InsertPacket(rtp_header, incoming_payload, payload_length);
+  return receiver_.InsertPacket(
+      rtp_header,
+      rtc::ArrayView<const uint8_t>(incoming_payload, payload_length));
 }
 
 // Minimum playout delay (Used for lip-sync).
 int AudioCodingModuleImpl::SetMinimumPlayoutDelay(int time_ms) {
   if ((time_ms < 0) || (time_ms > 10000)) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                  "Delay must be in the range of 0-1000 milliseconds.");
     return -1;
   }
   return receiver_.SetMinimumDelay(time_ms);
@@ skipping 158 matching lines @@
   return receiver_.LeastRequiredDelayMs();
 }
 
 void AudioCodingModuleImpl::GetDecodingCallStatistics(
       AudioDecodingCallStats* call_stats) const {
   receiver_.GetDecodingCallStatistics(call_stats);
 }
 
 }  // namespace acm2
 }  // namespace webrtc