RTCOutboundRTPStreamStats.roundTripTime: Only report non-negative values.

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 1610 matching lines @@
 
   ASSERT(report->Get(expected_video.id()));
   const RTCOutboundRTPStreamStats& video = report->Get(
       expected_video.id())->cast_to<RTCOutboundRTPStreamStats>();
   EXPECT_EQ(video, expected_video);
 
   EXPECT_TRUE(report->Get(*expected_video.transport_id));
   EXPECT_TRUE(report->Get(*expected_video.codec_id));
 }
 
+TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Default) {
+  MockVoiceMediaChannel* voice_media_channel = new MockVoiceMediaChannel();
+  cricket::VoiceChannel voice_channel(
+      test_->worker_thread(), test_->network_thread(), test_->media_engine(),
+      voice_media_channel, nullptr, "VoiceContentName", kDefaultRtcpEnabled,
+      kDefaultSrtpRequired);
+  MockVideoMediaChannel* video_media_channel = new MockVideoMediaChannel();
+  cricket::VideoChannel video_channel(
+      test_->worker_thread(), test_->network_thread(), video_media_channel,
+      nullptr, "VideoContentName", kDefaultRtcpEnabled, kDefaultSrtpRequired);
+
+  cricket::VoiceMediaInfo voice_media_info;
+  voice_media_info.senders.push_back(cricket::VoiceSenderInfo());
+  voice_media_info.senders[0].local_stats.push_back(cricket::SsrcSenderInfo());
+  voice_media_info.senders[0].local_stats[0].ssrc = 1;
+  voice_media_info.senders[0].packets_sent = 2;
+  voice_media_info.senders[0].bytes_sent = 3;
+  voice_media_info.senders[0].rtt_ms = -1;
+  voice_media_info.senders[0].codec_payload_type = rtc::Optional<int>(42);
+
+  cricket::VideoMediaInfo video_media_info;
+  video_media_info.senders.push_back(cricket::VideoSenderInfo());
+  video_media_info.senders[0].local_stats.push_back(cricket::SsrcSenderInfo());
+  video_media_info.senders[0].local_stats[0].ssrc = 1;
+  video_media_info.senders[0].firs_rcvd = 2;
+  video_media_info.senders[0].plis_rcvd = 3;
+  video_media_info.senders[0].nacks_rcvd = 4;
+  video_media_info.senders[0].packets_sent = 5;
+  video_media_info.senders[0].bytes_sent = 6;
+  video_media_info.senders[0].rtt_ms = -1;
+  video_media_info.senders[0].codec_payload_type = rtc::Optional<int>(42);
+
+  EXPECT_CALL(*voice_media_channel, GetStats(_))
+      .WillOnce(DoAll(SetArgPointee<0>(voice_media_info), Return(true)));
+  EXPECT_CALL(*video_media_channel, GetStats(_))
+      .WillOnce(DoAll(SetArgPointee<0>(video_media_info), Return(true)));
+
+  SessionStats session_stats;
+  session_stats.proxy_to_transport["VoiceContentName"] = "TransportName";
+  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(), voice_channel())
+      .WillRepeatedly(Return(&voice_channel));
+  EXPECT_CALL(test_->session(), video_channel())
+      .WillRepeatedly(Return(&video_channel));
+
+  rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
+
+  RTCOutboundRTPStreamStats expected_audio(
+      "RTCOutboundRTPAudioStream_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_OutboundAudio_42";
+  expected_audio.packets_sent = 2;
+  expected_audio.bytes_sent = 3;
+  // |expected_audio.round_trip_time| should be undefined.
+
+  ASSERT(report->Get(expected_audio.id()));
+  const RTCOutboundRTPStreamStats& audio = report->Get(
+      expected_audio.id())->cast_to<RTCOutboundRTPStreamStats>();
+  EXPECT_EQ(audio, expected_audio);
+
+  RTCOutboundRTPStreamStats expected_video(
+      "RTCOutboundRTPVideoStream_1", report->timestamp_us());
+  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_video.codec_id = "RTCCodec_OutboundVideo_42";
+  expected_video.fir_count = 2;
+  expected_video.pli_count = 3;
+  expected_video.nack_count = 4;
+  expected_video.packets_sent = 5;
+  expected_video.bytes_sent = 6;
+  // |expected_video.round_trip_time| should be undefined.
+
+  ASSERT(report->Get(expected_video.id()));
+  const RTCOutboundRTPStreamStats& video = report->Get(
+      expected_video.id())->cast_to<RTCOutboundRTPStreamStats>();
+  EXPECT_EQ(video, expected_video);
+}
+
 TEST_F(RTCStatsCollectorTest, CollectRTCTransportStats) {
   std::unique_ptr<cricket::Candidate> rtp_local_candidate = CreateFakeCandidate(
       "42.42.42.42", 42, "protocol", cricket::LOCAL_PORT_TYPE, 42);
   std::unique_ptr<cricket::Candidate> rtp_remote_candidate =
       CreateFakeCandidate("42.42.42.42", 42, "protocol",
                           cricket::LOCAL_PORT_TYPE, 42);
   std::unique_ptr<cricket::Candidate> rtcp_local_candidate =
       CreateFakeCandidate("42.42.42.42", 42, "protocol",
                           cricket::LOCAL_PORT_TYPE, 42);
   std::unique_ptr<cricket::Candidate> rtcp_remote_candidate =
@@ skipping 158 matching lines @@
   rtc::scoped_refptr<FakeRTCStatsCollector> collector_;
 };
 
 TEST_F(RTCStatsCollectorTestWithFakeCollector, ThreadUsageAndResultsMerging) {
   collector_->VerifyThreadUsageAndResultsMerging();
 }
 
 }  // namespace
 
 }  // namespace webrtc