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Populate “fractionLost” stats for remote inbound rtp streams

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Tests:
./out/Default/peerconnection_unittests

Manually tested with Chromium to see the data populated

Spec: https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict*
Bug: webrtc:12506
Change-Id: I60ef8061fb31deab06ca5f115246ceb5a8cdc5ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208960
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33361}
This commit is contained in:
Di Wu
2021-02-28 23:36:03 -08:00
committed by Commit Bot
parent 6512af04ba
commit 86f04ad135
5 changed files with 14 additions and 5 deletions
Download Patch File Download Diff File Expand all files Collapse all files

3
api/stats/rtcstats_objects.h
View File

@ -555,8 +555,7 @@ class RTC_EXPORT RTCRemoteInboundRtpStreamStats final : public RTCStats {
// RTCRemoteInboundRtpStreamStats
RTCStatsMember<std::string> local_id;
RTCStatsMember<double> round_trip_time;
// TODO(hbos): The following RTCRemoteInboundRtpStreamStats metric should also
// be implemented: fractionLost.
RTCStatsMember<double> fraction_lost;
};
// https://w3c.github.io/webrtc-stats/#dom-rtcmediasourcestats

2
pc/rtc_stats_collector.cc
View File

@ -541,6 +541,8 @@ ProduceRemoteInboundRtpStreamStatsFromReportBlockData(
remote_inbound->kind =
media_type == cricket::MEDIA_TYPE_AUDIO ? "audio" : "video";
remote_inbound->packets_lost = report_block.packets_lost;
remote_inbound->fraction_lost =
static_cast<double>(report_block.fraction_lost) / (1 << 8);
remote_inbound->round_trip_time =
static_cast<double>(report_block_data.last_rtt_ms()) /
rtc::kNumMillisecsPerSec;

4
pc/rtc_stats_collector_unittest.cc
View File

@ -2678,6 +2678,7 @@ TEST_P(RTCStatsCollectorTestWithParamKind,
const int64_t kReportBlockTimestampUtcUs = 123456789;
const int64_t kRoundTripTimeMs = 13000;
const double kRoundTripTimeSeconds = 13.0;
const uint8_t kFractionLost = 12;
// The report block's timestamp cannot be from the future, set the fake clock
// to match.
@ -2690,6 +2691,7 @@ TEST_P(RTCStatsCollectorTestWithParamKind,
// |source_ssrc|, "SSRC of the RTP packet sender".
report_block.source_ssrc = ssrc;
report_block.packets_lost = 7;
report_block.fraction_lost = kFractionLost;
ReportBlockData report_block_data;
report_block_data.SetReportBlock(report_block, kReportBlockTimestampUtcUs);
report_block_data.AddRoundTripTimeSample(1234);
@ -2708,6 +2710,8 @@ TEST_P(RTCStatsCollectorTestWithParamKind,
"RTCRemoteInboundRtp" + MediaTypeUpperCase() + stream_id,
kReportBlockTimestampUtcUs);
expected_remote_inbound_rtp.ssrc = ssrc;
expected_remote_inbound_rtp.fraction_lost =
static_cast<double>(kFractionLost) / (1 << 8);
expected_remote_inbound_rtp.kind = MediaTypeLowerCase();
expected_remote_inbound_rtp.transport_id =
"RTCTransport_TransportName_1"; // 1 for RTP (we have no RTCP

1
pc/rtc_stats_integrationtest.cc
View File

@ -1014,6 +1014,7 @@ class RTCStatsReportVerifier {
verifier.TestMemberIsIDReference(remote_inbound_stream.codec_id,
RTCCodecStats::kType);
verifier.TestMemberIsDefined(remote_inbound_stream.packets_lost);
verifier.TestMemberIsDefined(remote_inbound_stream.fraction_lost);
// Note that the existance of RTCCodecStats is needed for |codec_id| and
// |jitter| to be present.
verifier.TestMemberIsNonNegative<double>(remote_inbound_stream.jitter);

9
stats/rtcstats_objects.cc
View File

@ -848,7 +848,8 @@ WEBRTC_RTCSTATS_IMPL(
&packets_lost,
&jitter,
&local_id,
&round_trip_time)
&round_trip_time,
&fraction_lost)
// clang-format on
RTCRemoteInboundRtpStreamStats::RTCRemoteInboundRtpStreamStats(
@ -867,7 +868,8 @@ RTCRemoteInboundRtpStreamStats::RTCRemoteInboundRtpStreamStats(
packets_lost("packetsLost"),
jitter("jitter"),
local_id("localId"),
round_trip_time("roundTripTime") {}
round_trip_time("roundTripTime"),
fraction_lost("fractionLost") {}
RTCRemoteInboundRtpStreamStats::RTCRemoteInboundRtpStreamStats(
const RTCRemoteInboundRtpStreamStats& other)
@ -879,7 +881,8 @@ RTCRemoteInboundRtpStreamStats::RTCRemoteInboundRtpStreamStats(
packets_lost(other.packets_lost),
jitter(other.jitter),
local_id(other.local_id),
round_trip_time(other.round_trip_time) {}
round_trip_time(other.round_trip_time),
fraction_lost(other.fraction_lost) {}
RTCRemoteInboundRtpStreamStats::~RTCRemoteInboundRtpStreamStats() {}