Renamed fields in common_types.h/RtcpStatistics.

webrtc/modules/audio_coding/neteq/neteq_unittest.cc

 /*
  *  Copyright (c) 2011 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 74 matching lines @@
   stats->set_added_zero_samples(stats_raw.added_zero_samples);
   stats->set_mean_waiting_time_ms(stats_raw.mean_waiting_time_ms);
   stats->set_median_waiting_time_ms(stats_raw.median_waiting_time_ms);
   stats->set_min_waiting_time_ms(stats_raw.min_waiting_time_ms);
   stats->set_max_waiting_time_ms(stats_raw.max_waiting_time_ms);
 }
 
 void Convert(const webrtc::RtcpStatistics& stats_raw,
              webrtc::neteq_unittest::RtcpStatistics* stats) {
   stats->set_fraction_lost(stats_raw.fraction_lost);
-  stats->set_cumulative_lost(stats_raw.cumulative_lost);
+  stats->set_cumulative_lost(stats_raw.packets_lost);
   stats->set_extended_max_sequence_number(
-      stats_raw.extended_max_sequence_number);
+      stats_raw.extended_highest_sequence_number);
   stats->set_jitter(stats_raw.jitter);
 }
 
 void AddMessage(FILE* file, rtc::MessageDigest* digest,
                 const std::string& message) {
   int32_t size = message.length();
   if (file)
     ASSERT_EQ(1u, fwrite(&size, sizeof(size), 1, file));
   digest->Update(&size, sizeof(size));
 
@@ skipping 1520 matching lines @@
   // Pull out data once.
   AudioFrame output;
   bool muted;
   ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
 
   EXPECT_EQ(std::vector<uint32_t>({kRtpTimestamp1, kRtpTimestamp2}),
             neteq_->LastDecodedTimestamps());
 }
 
 }  // namespace webrtc