RTCInboundRTPStreamStats.packetsLost set by RTCStatsCollector.

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 1293 matching lines @@
   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.receivers.push_back(cricket::VoiceReceiverInfo());
   voice_media_info.receivers[0].local_stats.push_back(
       cricket::SsrcReceiverInfo());
   voice_media_info.receivers[0].local_stats[0].ssrc = 1;
+  voice_media_info.receivers[0].packets_lost = 42;
   voice_media_info.receivers[0].packets_rcvd = 2;
   voice_media_info.receivers[0].bytes_rcvd = 3;
   voice_media_info.receivers[0].codec_payload_type = rtc::Optional<int>(42);
   voice_media_info.receivers[0].jitter_ms = 4500;
   voice_media_info.receivers[0].fraction_lost = 5.5f;
 
   RtpCodecParameters codec_parameters;
   codec_parameters.payload_type = 42;
   codec_parameters.mime_type = "dummy";
   codec_parameters.clock_rate = 0;
@@ skipping 24 matching lines @@
   RTCInboundRTPStreamStats expected_audio(
       "RTCInboundRTPAudioStream_1", report->timestamp_us());
   expected_audio.ssrc = "1";
   expected_audio.is_remote = false;
   expected_audio.media_type = "audio";
   expected_audio.transport_id = "RTCTransport_TransportName_" +
       rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP);
   expected_audio.codec_id = "RTCCodec_InboundAudio_42";
   expected_audio.packets_received = 2;
   expected_audio.bytes_received = 3;
+  expected_audio.packets_lost = 42;
   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));
   EXPECT_TRUE(report->Get(*expected_audio.codec_id));
 }
 
 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].packets_lost = 42;
   video_media_info.receivers[0].bytes_rcvd = 3;
   video_media_info.receivers[0].fraction_lost = 4.5f;
   video_media_info.receivers[0].codec_payload_type = rtc::Optional<int>(42);
   video_media_info.receivers[0].firs_sent = 5;
   video_media_info.receivers[0].plis_sent = 6;
   video_media_info.receivers[0].nacks_sent = 7;
 
   RtpCodecParameters codec_parameters;
   codec_parameters.payload_type = 42;
   codec_parameters.mime_type = "dummy";
@@ skipping 28 matching lines @@
   expected_video.is_remote = false;
   expected_video.media_type = "video";
   expected_video.transport_id = "RTCTransport_TransportName_" +
       rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP);
   expected_video.codec_id = "RTCCodec_InboundVideo_42";
   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.packets_lost = 42;
   expected_video.fraction_lost = 4.5;
 
   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_video.transport_id));
   EXPECT_TRUE(report->Get(*video.codec_id));
 }
@@ skipping 305 matching lines @@
   rtc::scoped_refptr<FakeRTCStatsCollector> collector_;
 };
 
 TEST_F(RTCStatsCollectorTestWithFakeCollector, ThreadUsageAndResultsMerging) {
   collector_->VerifyThreadUsageAndResultsMerging();
 }
 
 }  // namespace
 
 }  // namespace webrtc