Change NetEq::InsertPacket to take an RTPHeader

webrtc/modules/audio_coding/acm2/acm_receiver.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 86 matching lines @@
         return 0;
       }
     } else {
       last_audio_decoder_ = ci;
       last_audio_format_ = neteq_->GetDecoderFormat(ci->pltype);
       RTC_DCHECK(last_audio_format_);
       last_packet_sample_rate_hz_ = rtc::Optional<int>(ci->plfreq);
     }
   }  // |crit_sect_| is released.
 
-  if (neteq_->InsertPacket(rtp_header, incoming_payload, receive_timestamp) <
-      0) {
+  if (neteq_->InsertPacket(rtp_header.header, incoming_payload,
+                           receive_timestamp) < 0) {
     LOG(LERROR) << "AcmReceiver::InsertPacket "
                 << static_cast<int>(header->payloadType)
                 << " Failed to insert packet";
     return -1;
   }
   return 0;
 }
 
 int AcmReceiver::GetAudio(int desired_freq_hz,
                           AudioFrame* audio_frame,
@@ skipping 272 matching lines @@
 
 void AcmReceiver::GetDecodingCallStatistics(
     AudioDecodingCallStats* stats) const {
   rtc::CritScope lock(&crit_sect_);
   *stats = call_stats_.GetDecodingStatistics();
 }
 
 }  // namespace acm2
 
 }  // namespace webrtc