Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
The value is being moved:
https://github.com/w3c/webrtc-stats/pull/167
Stop collecting this value. Our previous value was incorrect, our RTT
value was a smoothed value based on STUN pings but the spec says it
should be based on RTCP timestamps in RTCP Receiver Report (RR) on
inbound streams with isRemote=true (not supported).
Updated some bug references.
BUG=webrtc:7065, webrtc:7066
Review-Url: https://codereview.webrtc.org/2722633005
Cr-Original-Commit-Position: refs/heads/master@{#16931}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: a7a9be159df454db9d861b49b94b3d02a283eb1c
diff --git a/pc/rtcstatscollector_unittest.cc b/pc/rtcstatscollector_unittest.cc
index db26096..ea7b34a 100644
--- a/pc/rtcstatscollector_unittest.cc
+++ b/pc/rtcstatscollector_unittest.cc
@@ -1959,7 +1959,6 @@
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);
RtpCodecParameters codec_parameters;
@@ -2005,22 +2004,12 @@
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_TRUE(report->Get(expected_audio.id()));
EXPECT_EQ(
report->Get(expected_audio.id())->cast_to<RTCOutboundRTPStreamStats>(),
expected_audio);
- // Set previously undefined values and "GetStats" again.
- voice_media_info.senders[0].rtt_ms = 4500;
- expected_audio.round_trip_time = 4.5;
-
- EXPECT_CALL(*voice_media_channel, GetStats(_))
- .WillOnce(DoAll(SetArgPointee<0>(voice_media_info), Return(true)));
- collector_->ClearCachedStatsReport();
- report = GetStatsReport();
-
ASSERT_TRUE(report->Get(expected_audio.id()));
EXPECT_EQ(
report->Get(expected_audio.id())->cast_to<RTCOutboundRTPStreamStats>(),
@@ -2050,7 +2039,6 @@
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);
video_media_info.senders[0].frames_encoded = 8;
video_media_info.senders[0].qp_sum = rtc::Optional<uint64_t>();
@@ -2102,8 +2090,7 @@
expected_video.packets_sent = 5;
expected_video.bytes_sent = 6;
expected_video.frames_encoded = 8;
- // |expected_video.round_trip_time| and |expected_video.qp_sum| should be
- // undefined.
+ // |expected_video.qp_sum| should be undefined.
ASSERT_TRUE(report->Get(expected_video.id()));
EXPECT_EQ(
@@ -2111,9 +2098,7 @@
expected_video);
// Set previously undefined values and "GetStats" again.
- video_media_info.senders[0].rtt_ms = 7500;
video_media_info.senders[0].qp_sum = rtc::Optional<uint64_t>(9);
- expected_video.round_trip_time = 7.5;
expected_video.qp_sum = 9;
EXPECT_CALL(*video_media_channel, GetStats(_))
