Change NetEq::InsertPacket to take an RTPHeader

webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.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 192 matching lines @@
     for (size_t i = 0; i < output_.samples_per_channel_; ++i) {
       ASSERT_EQ(output_.data_[i], output_internal_.data_[i])
           << "Diff in sample " << i << ".";
     }
   }
 
   void InsertPacket(WebRtcRTPHeader rtp_header,
                     rtc::ArrayView<const uint8_t> payload,
                     uint32_t receive_timestamp) override {
     // Insert packet in internal decoder.
-    ASSERT_EQ(NetEq::kOK, neteq_internal_->InsertPacket(rtp_header, payload,
-                                                        receive_timestamp));
+    ASSERT_EQ(NetEq::kOK, neteq_internal_->InsertPacket(
+                              rtp_header.header, payload, receive_timestamp));
 
     // Insert packet in external decoder instance.
     NetEqExternalDecoderUnitTest::InsertPacket(rtp_header, payload,
                                                receive_timestamp);
   }
 
   int NumExpectedDecodeCalls(int num_loops) override { return num_loops; }
 
  private:
   int sample_rate_hz_;
@@ skipping 220 matching lines @@
                                                         kStartSeqeunceNumber,
                                                         kStartTimestamp,
                                                         kJumpFromTimestamp,
                                                         kJumpToTimestamp));
 
   RunTest(130);  // Run 130 laps @ 10 ms each in the test loop.
   EXPECT_EQ(kRecovered, test_state_);
 }
 
 }  // namespace webrtc