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

webrtc/api/rtcstatscollector_unittest.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 1365 matching lines @@
       nullptr, "VideoContentName", false);
 
   cricket::VideoMediaInfo video_media_info;
   video_media_info.receivers.push_back(cricket::VideoReceiverInfo());
   video_media_info.receivers[0].local_stats.push_back(
       cricket::SsrcReceiverInfo());
   video_media_info.receivers[0].local_stats[0].ssrc = 1;
   video_media_info.receivers[0].packets_rcvd = 2;
   video_media_info.receivers[0].bytes_rcvd = 3;
   video_media_info.receivers[0].fraction_lost = 4.5f;
+  video_media_info.receivers[0].firs_sent = 5;
+  video_media_info.receivers[0].plis_sent = 6;
+  video_media_info.receivers[0].nacks_sent = 7;
   EXPECT_CALL(*video_media_channel, GetStats(_))
       .WillOnce(DoAll(SetArgPointee<0>(video_media_info), Return(true)));
 
   SessionStats session_stats;
   session_stats.proxy_to_transport["VideoContentName"] = "TransportName";
   session_stats.transport_stats["TransportName"].transport_name =
       "TransportName";
 
   // Make sure the associated |RTCTransportStats| is created.
   cricket::TransportChannelStats channel_stats;
   channel_stats.component = cricket::ICE_CANDIDATE_COMPONENT_RTP;
   session_stats.transport_stats["TransportName"].channel_stats.push_back(
       channel_stats);
 
   EXPECT_CALL(test_->session(), GetTransportStats(_))
       .WillRepeatedly(DoAll(SetArgPointee<0>(session_stats), Return(true)));
   EXPECT_CALL(test_->session(), video_channel())
       .WillRepeatedly(Return(&video_channel));
 
   rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
 
-  RTCInboundRTPStreamStats expected_audio(
+  RTCInboundRTPStreamStats expected_video(
       "RTCInboundRTPVideoStream_1", report->timestamp_us());
-  expected_audio.ssrc = "1";
-  expected_audio.is_remote = false;
-  expected_audio.media_type = "video";
-  expected_audio.transport_id = "RTCTransport_TransportName_" +
+  expected_video.ssrc = "1";
+  expected_video.is_remote = false;
+  expected_video.media_type = "video";
+  expected_video.transport_id = "RTCTransport_TransportName_" +
       rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP);
-  expected_audio.packets_received = 2;
-  expected_audio.bytes_received = 3;
-  expected_audio.fraction_lost = 4.5;
+  expected_video.fir_count = 5;
+  expected_video.pli_count = 6;
+  expected_video.nack_count = 7;
+  expected_video.packets_received = 2;
+  expected_video.bytes_received = 3;
+  expected_video.fraction_lost = 4.5;
 
-  ASSERT(report->Get(expected_audio.id()));
-  const RTCInboundRTPStreamStats& audio = report->Get(
-      expected_audio.id())->cast_to<RTCInboundRTPStreamStats>();
-  EXPECT_EQ(audio, expected_audio);
+  ASSERT(report->Get(expected_video.id()));
+  const RTCInboundRTPStreamStats& video = report->Get(
+      expected_video.id())->cast_to<RTCInboundRTPStreamStats>();
+  EXPECT_EQ(video, expected_video);
 
-  EXPECT_TRUE(report->Get(*expected_audio.transport_id));
+  EXPECT_TRUE(report->Get(*expected_video.transport_id));
 }
 
 TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) {
   MockVoiceMediaChannel* voice_media_channel = new MockVoiceMediaChannel();
   cricket::VoiceChannel voice_channel(
       test_->worker_thread(), test_->network_thread(), test_->media_engine(),
       voice_media_channel, nullptr, "VoiceContentName", false);
 
   cricket::VoiceMediaInfo voice_media_info;
   voice_media_info.senders.push_back(cricket::VoiceSenderInfo());
@@ skipping 213 matching lines @@
   rtc::scoped_refptr<FakeRTCStatsCollector> collector_;
 };
 
 TEST_F(RTCStatsCollectorTestWithFakeCollector, ThreadUsageAndResultsMerging) {
   collector_->VerifyThreadUsageAndResultsMerging();
 }
 
 }  // namespace
 
 }  // namespace webrtc