Implement nack_count metric for inbound audio rtp streams.

Bug: webrtc:12925
Change-Id: I4542ca0f14a7dd7485ad5a2b6f2bd7051076f71f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224085
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34401}
diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc
index 1fa731b..070400c 100644
--- a/pc/rtc_stats_collector.cc
+++ b/pc/rtc_stats_collector.cc
@@ -337,6 +337,9 @@
       media_receiver_info.jitter_buffer_delay_seconds;
   inbound_stats->jitter_buffer_emitted_count =
       media_receiver_info.jitter_buffer_emitted_count;
+  if (media_receiver_info.nacks_sent) {
+    inbound_stats->nack_count = *media_receiver_info.nacks_sent;
+  }
 }
 
 std::unique_ptr<RTCInboundRTPStreamStats> CreateInboundAudioStreamStats(
@@ -454,8 +457,6 @@
       static_cast<uint32_t>(video_receiver_info.firs_sent);
   inbound_video->pli_count =
       static_cast<uint32_t>(video_receiver_info.plis_sent);
-  inbound_video->nack_count =
-      static_cast<uint32_t>(video_receiver_info.nacks_sent);
   inbound_video->frames_received = video_receiver_info.frames_received;
   inbound_video->frames_decoded = video_receiver_info.frames_decoded;
   inbound_video->frames_dropped = video_receiver_info.frames_dropped;