RTCInboundRTPStreamStats's [fir/pli/nack]_count are collected for video.

webrtc/api/rtcstatscollector.cc

 /*
  *  Copyright 2016 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 89 matching lines @@
   }
 }
 
 void SetMediaStreamTrackStatsFromMediaStreamTrackInterface(
     const MediaStreamTrackInterface& track,
     RTCMediaStreamTrackStats* track_stats) {
   track_stats->track_identifier = track.id();
   track_stats->ended = (track.state() == MediaStreamTrackInterface::kEnded);
 }
 
 void SetInboundRTPStreamStatsFromMediaReceiverInfo(

[hta-webrtc, 2016/11/22 09:51:29]

Add a comment to this function to say that this provides the media independent
counters. It's obvious when you inspect it, but nice to be explicit.

[hbos, 2016/11/22 11:02:45]

Done.

     const cricket::MediaReceiverInfo& media_receiver_info,
     RTCInboundRTPStreamStats* inbound_stats) {
   RTC_DCHECK(inbound_stats);
   inbound_stats->ssrc = rtc::ToString<>(media_receiver_info.ssrc());
   // TODO(hbos): Support the remote case. crbug.com/657855
   inbound_stats->is_remote = false;
   // TODO(hbos): Set |codec_id| when we have |RTCCodecStats|. Maybe relevant:
   // |media_receiver_info.codec_name|. crbug.com/657854, 657855, 659117
   inbound_stats->packets_received =
       static_cast<uint32_t>(media_receiver_info.packets_rcvd);
   inbound_stats->bytes_received =
       static_cast<uint64_t>(media_receiver_info.bytes_rcvd);
   inbound_stats->fraction_lost =
       static_cast<double>(media_receiver_info.fraction_lost);
 }
 
 void SetInboundRTPStreamStatsFromVoiceReceiverInfo(
     const cricket::VoiceReceiverInfo& voice_receiver_info,
-    RTCInboundRTPStreamStats* inbound_stats) {
+    RTCInboundRTPStreamStats* inbound_audio) {
   SetInboundRTPStreamStatsFromMediaReceiverInfo(
-      voice_receiver_info, inbound_stats);
-  inbound_stats->media_type = "audio";
-  inbound_stats->jitter =
+      voice_receiver_info, inbound_audio);
+  inbound_audio->media_type = "audio";
+  inbound_audio->jitter =
       static_cast<double>(voice_receiver_info.jitter_ms) /
           rtc::kNumMillisecsPerSec;
+  // |fir_count|, |pli_count| and |sli_count| are only valid for video and are
+  // purposefully left undefined for audio.
 }
 
 void SetInboundRTPStreamStatsFromVideoReceiverInfo(
     const cricket::VideoReceiverInfo& video_receiver_info,
-    RTCInboundRTPStreamStats* inbound_stats) {
+    RTCInboundRTPStreamStats* inbound_video) {
   SetInboundRTPStreamStatsFromMediaReceiverInfo(
-      video_receiver_info, inbound_stats);
-  inbound_stats->media_type = "video";
+      video_receiver_info, inbound_video);
+  inbound_video->media_type = "video";
+  inbound_video->fir_count =
+      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);

[hta-webrtc, 2016/11/22 09:51:29]

Are we sure these are actually counted (ie never -1)?
If they're -1, we should skip generating the stat, but if they're never -1, we
don't need to bother.

[hbos, 2016/11/22 11:02:45]

Looks like it, they start at 0 and there is code for increasing them.

 }
 
 void SetOutboundRTPStreamStatsFromMediaSenderInfo(
     const cricket::MediaSenderInfo& media_sender_info,
     RTCOutboundRTPStreamStats* outbound_stats) {
   RTC_DCHECK(outbound_stats);
   outbound_stats->ssrc = rtc::ToString<>(media_sender_info.ssrc());
   // TODO(hbos): Support the remote case. crbug.com/657856
   outbound_stats->is_remote = false;
   // TODO(hbos): Set |codec_id| when we have |RTCCodecStats|. Maybe relevant:
@@ skipping 649 matching lines @@
     const std::string& type) {
   return CandidateTypeToRTCIceCandidateType(type);
 }
 
 const char* DataStateToRTCDataChannelStateForTesting(
     DataChannelInterface::DataState state) {
   return DataStateToRTCDataChannelState(state);
 }
 
 }  // namespace webrtc