RTCOutboundRTPStreamStats.roundTripTime: Only report non-negative values.
Underlying stats gatherers may otherwise default it to -1.
BUG=chromium:669877, chromium:627816
Review-Url: https://codereview.webrtc.org/2562703007
Cr-Commit-Position: refs/heads/master@{#15625}
diff --git a/webrtc/api/rtcstats_integrationtest.cc b/webrtc/api/rtcstats_integrationtest.cc
index 923997c..0f60377 100644
--- a/webrtc/api/rtcstats_integrationtest.cc
+++ b/webrtc/api/rtcstats_integrationtest.cc
@@ -497,7 +497,8 @@
verifier.TestMemberIsDefined(outbound_stream.packets_sent);
verifier.TestMemberIsDefined(outbound_stream.bytes_sent);
verifier.TestMemberIsUndefined(outbound_stream.target_bitrate);
- verifier.TestMemberIsDefined(outbound_stream.round_trip_time);
+ // TODO(hbos): Defined in video but not audio case. Why? crbug.com/669877
+ verifier.MarkMemberTested(outbound_stream.round_trip_time, true);
return verifier.ExpectAllMembersSuccessfullyTested();
}
diff --git a/webrtc/api/rtcstatscollector.cc b/webrtc/api/rtcstatscollector.cc
index 5289c48..f5d3e5f 100644
--- a/webrtc/api/rtcstatscollector.cc
+++ b/webrtc/api/rtcstatscollector.cc
@@ -197,8 +197,10 @@
static_cast<uint32_t>(media_sender_info.packets_sent);
outbound_stats->bytes_sent =
static_cast<uint64_t>(media_sender_info.bytes_sent);
- outbound_stats->round_trip_time =
- static_cast<double>(media_sender_info.rtt_ms) / rtc::kNumMillisecsPerSec;
+ if (media_sender_info.rtt_ms >= 0) {
+ outbound_stats->round_trip_time = static_cast<double>(
+ media_sender_info.rtt_ms) / rtc::kNumMillisecsPerSec;
+ }
}
void SetOutboundRTPStreamStatsFromVoiceSenderInfo(
diff --git a/webrtc/api/rtcstatscollector_unittest.cc b/webrtc/api/rtcstatscollector_unittest.cc
index 6d0b56f..0e1fef0 100644
--- a/webrtc/api/rtcstatscollector_unittest.cc
+++ b/webrtc/api/rtcstatscollector_unittest.cc
@@ -1628,6 +1628,102 @@
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);
