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 1348 matching lines @@
   expected_audio.packets_received = 2;
   expected_audio.bytes_received = 3;
   expected_audio.jitter = 4.5;
   expected_audio.fraction_lost = 5.5;
 
   ASSERT(report->Get(expected_audio.id()));
   const RTCInboundRTPStreamStats& audio = report->Get(
       expected_audio.id())->cast_to<RTCInboundRTPStreamStats>();
   EXPECT_EQ(audio, expected_audio);
 
   EXPECT_TRUE(report->Get(*expected_audio.transport_id));

[hta-webrtc, 2016/11/22 09:51:30]

Can you add a test line here that checks that fir_count is not set?

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

Because we do EXPECT_EQ between expected and actual RTCStats, and the expected
stats does not have these members set, the test already makes sure that they are
not set. Same for the other tests.

 }
 
 TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
   MockVideoMediaChannel* video_media_channel = new MockVideoMediaChannel();
   cricket::VideoChannel video_channel(
       test_->worker_thread(), test_->network_thread(), video_media_channel,
       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