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-Commit-Position: refs/heads/master@{#16931}
diff --git a/webrtc/api/stats/rtcstats_objects.h b/webrtc/api/stats/rtcstats_objects.h
index cc01b5e..49135f8 100644
--- a/webrtc/api/stats/rtcstats_objects.h
+++ b/webrtc/api/stats/rtcstats_objects.h
@@ -327,8 +327,8 @@
};
// https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*
-// Tracking bug crbug.com/657855
-// TODO(hbos): Support the remote case |is_remote = true|. crbug.com/657855
+// TODO(hbos): Support the remote case |is_remote = true|.
+// https://bugs.webrtc.org/7065
class RTCInboundRTPStreamStats final : public RTCRTPStreamStats {
public:
WEBRTC_RTCSTATS_DECL();
@@ -341,36 +341,38 @@
RTCStatsMember<uint32_t> packets_received;
RTCStatsMember<uint64_t> bytes_received;
RTCStatsMember<uint32_t> packets_lost;
- // TODO(hbos): Not collected in the "video" case by |RTCStatsCollector|.
- // crbug.com/657855
+ // TODO(hbos): Collect and populate this value for both "audio" and "video",
+ // currently not collected for "video". https://bugs.webrtc.org/7065
RTCStatsMember<double> jitter;
RTCStatsMember<double> fraction_lost;
- // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+ // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
+ RTCStatsMember<double> round_trip_time;
+ // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
RTCStatsMember<uint32_t> packets_discarded;
- // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+ // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
RTCStatsMember<uint32_t> packets_repaired;
- // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+ // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
RTCStatsMember<uint32_t> burst_packets_lost;
- // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+ // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
RTCStatsMember<uint32_t> burst_packets_discarded;
- // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+ // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
RTCStatsMember<uint32_t> burst_loss_count;
- // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+ // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
RTCStatsMember<uint32_t> burst_discard_count;
- // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+ // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
RTCStatsMember<double> burst_loss_rate;
- // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+ // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
RTCStatsMember<double> burst_discard_rate;
- // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+ // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
RTCStatsMember<double> gap_loss_rate;
- // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+ // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
RTCStatsMember<double> gap_discard_rate;
RTCStatsMember<uint32_t> frames_decoded;
};
// https://w3c.github.io/webrtc-stats/#outboundrtpstats-dict*
-// Tracking bug crbug.com/657856
-// TODO(hbos): Support the remote case |is_remote = true|. crbug.com/657856
+// TODO(hbos): Support the remote case |is_remote = true|.
+// https://bugs.webrtc.org/7066
class RTCOutboundRTPStreamStats final : public RTCRTPStreamStats {
public:
WEBRTC_RTCSTATS_DECL();
@@ -382,9 +384,8 @@
RTCStatsMember<uint32_t> packets_sent;
RTCStatsMember<uint64_t> bytes_sent;
- // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657856
+ // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7066
RTCStatsMember<double> target_bitrate;
- RTCStatsMember<double> round_trip_time;
RTCStatsMember<uint32_t> frames_encoded;
};
diff --git a/webrtc/pc/rtcstats_integrationtest.cc b/webrtc/pc/rtcstats_integrationtest.cc
index 1176b54..922e07c 100644
--- a/webrtc/pc/rtcstats_integrationtest.cc
+++ b/webrtc/pc/rtcstats_integrationtest.cc
@@ -553,6 +553,7 @@
verifier.TestMemberIsNonNegative<double>(inbound_stream.jitter);
}
verifier.TestMemberIsNonNegative<double>(inbound_stream.fraction_lost);
+ verifier.TestMemberIsUndefined(inbound_stream.round_trip_time);
verifier.TestMemberIsUndefined(inbound_stream.packets_discarded);
verifier.TestMemberIsUndefined(inbound_stream.packets_repaired);
verifier.TestMemberIsUndefined(inbound_stream.burst_packets_lost);
@@ -585,8 +586,6 @@
verifier.TestMemberIsNonNegative<uint32_t>(outbound_stream.packets_sent);
verifier.TestMemberIsNonNegative<uint64_t>(outbound_stream.bytes_sent);
verifier.TestMemberIsUndefined(outbound_stream.target_bitrate);
- // TODO(hbos): Defined in video but not audio case. Why? crbug.com/669877
- verifier.MarkMemberTested(outbound_stream.round_trip_time, true);
if (outbound_stream.media_type.is_defined() &&
*outbound_stream.media_type == "video") {
verifier.TestMemberIsDefined(outbound_stream.frames_encoded);
diff --git a/webrtc/pc/rtcstatscollector.cc b/webrtc/pc/rtcstatscollector.cc
index 36a4bde..347bca5 100644
--- a/webrtc/pc/rtcstatscollector.cc
+++ b/webrtc/pc/rtcstatscollector.cc
@@ -261,10 +261,6 @@
static_cast<uint32_t>(media_sender_info.packets_sent);
outbound_stats->bytes_sent =
static_cast<uint64_t>(media_sender_info.bytes_sent);
- 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/pc/rtcstatscollector_unittest.cc b/webrtc/pc/rtcstatscollector_unittest.cc
index db26096..ea7b34a 100644
--- a/webrtc/pc/rtcstatscollector_unittest.cc
+++ b/webrtc/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(_))
diff --git a/webrtc/stats/rtcstats_objects.cc b/webrtc/stats/rtcstats_objects.cc
index 9a7ebed..5397ef7 100644
--- a/webrtc/stats/rtcstats_objects.cc
+++ b/webrtc/stats/rtcstats_objects.cc
@@ -513,6 +513,7 @@
&packets_lost,
&jitter,
&fraction_lost,
+ &round_trip_time,
&packets_discarded,
&packets_repaired,
&burst_packets_lost,
@@ -538,6 +539,7 @@
packets_lost("packetsLost"),
jitter("jitter"),
fraction_lost("fractionLost"),
+ round_trip_time("roundTripTime"),
packets_discarded("packetsDiscarded"),
packets_repaired("packetsRepaired"),
burst_packets_lost("burstPacketsLost"),
@@ -559,6 +561,7 @@
packets_lost(other.packets_lost),
jitter(other.jitter),
fraction_lost(other.fraction_lost),
+ round_trip_time(other.round_trip_time),
packets_discarded(other.packets_discarded),
packets_repaired(other.packets_repaired),
burst_packets_lost(other.burst_packets_lost),
@@ -580,7 +583,6 @@
&packets_sent,
&bytes_sent,
&target_bitrate,
- &round_trip_time,
&frames_encoded);
RTCOutboundRTPStreamStats::RTCOutboundRTPStreamStats(
@@ -594,7 +596,6 @@
packets_sent("packetsSent"),
bytes_sent("bytesSent"),
target_bitrate("targetBitrate"),
- round_trip_time("roundTripTime"),
frames_encoded("framesEncoded") {
}
@@ -604,7 +605,6 @@
packets_sent(other.packets_sent),
bytes_sent(other.bytes_sent),
target_bitrate(other.target_bitrate),
- round_trip_time(other.round_trip_time),
frames_encoded(other.frames_encoded) {
}
