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Move more non-standard metrics to inbound-rtp.

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They may be non-standard, but they shouldn't be on a stats dictionary
that is deprecated (track is going away soon-ish). By moving them to
inbound-rtp they can continue to exist beyond track deprecation and
live in the right place in case we decide to standardize them later.

To help downstream projects transitions, the metrics are temporarily
available in both old and new locations. Delete of old location will
happen in a follow-up CL. TODOs added.

Bug: webrtc:14524
Change-Id: I2008060fa4ba76cde859d9144d2bb9648c7ff9af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278200
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38315}
This commit is contained in:
Henrik Boström 2022-10-06 13:37:11 +02:00 committed by WebRTC LUCI CQ
parent 8ad5e393c4
commit 2fb83072db
6 changed files with 96 additions and 23 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

11
api/stats/rtcstats_objects.h
View file

@ -346,13 +346,10 @@ class RTC_EXPORT RTCMediaStreamTrackStats final : public RTCStats {
RTCStatsMember<uint64_t> concealment_events;
RTCStatsMember<uint64_t> inserted_samples_for_deceleration;
RTCStatsMember<uint64_t> removed_samples_for_acceleration;
// Non-standard audio-only member
// https://w3c.github.io/webrtc-provisional-stats/#dom-rtcaudioreceiverstats-jitterbufferflushes
// TODO(crbug.com/webrtc/14524): These metrics have been moved, delete them.
RTCNonStandardStatsMember<uint64_t> jitter_buffer_flushes;
RTCNonStandardStatsMember<uint64_t> delayed_packet_outage_samples;
RTCNonStandardStatsMember<double> relative_packet_arrival_delay;
// TODO(henrik.lundin): Add description of the interruption metrics at
// https://github.com/w3c/webrtc-provisional-stats/issues/17
RTCNonStandardStatsMember<uint32_t> interruption_count;
RTCNonStandardStatsMember<double> total_interruption_duration;
// Non-standard video-only members.
@ -503,6 +500,12 @@ class RTC_EXPORT RTCInboundRTPStreamStats final
RTCStatsMember<uint32_t> pli_count;
RTCStatsMember<uint32_t> nack_count;
RTCStatsMember<uint64_t> qp_sum;
// Non-standard audio metrics.
RTCNonStandardStatsMember<uint64_t> jitter_buffer_flushes;
RTCNonStandardStatsMember<uint64_t> delayed_packet_outage_samples;
RTCNonStandardStatsMember<double> relative_packet_arrival_delay;
RTCNonStandardStatsMember<uint32_t> interruption_count;
RTCNonStandardStatsMember<double> total_interruption_duration;
// The former googMinPlayoutDelayMs (in seconds).
RTCNonStandardStatsMember<double> min_playout_delay;

17
pc/rtc_stats_collector.cc
View file

@ -496,6 +496,19 @@ std::unique_ptr<RTCInboundRTPStreamStats> CreateInboundAudioStreamStats(
inbound_audio->fec_packets_discarded =
voice_receiver_info.fec_packets_discarded;
inbound_audio->packets_discarded = voice_receiver_info.packets_discarded;
inbound_audio->jitter_buffer_flushes =
voice_receiver_info.jitter_buffer_flushes;
inbound_audio->delayed_packet_outage_samples =
voice_receiver_info.delayed_packet_outage_samples;
inbound_audio->relative_packet_arrival_delay =
voice_receiver_info.relative_packet_arrival_delay_seconds;
inbound_audio->interruption_count =
voice_receiver_info.interruption_count >= 0
? voice_receiver_info.interruption_count
: 0;
inbound_audio->total_interruption_duration =
static_cast<double>(voice_receiver_info.total_interruption_duration_ms) /
rtc::kNumMillisecsPerSec;
return inbound_audio;
}
@ -1034,6 +1047,9 @@ ProduceMediaStreamTrackStatsFromVoiceReceiverInfo(
voice_receiver_info.silent_concealed_samples;
audio_track_stats->concealment_events =
voice_receiver_info.concealment_events;
// TODO(crbug.com/webrtc/14524): These metrics have been moved from "track"
// stats, delete them.
audio_track_stats->jitter_buffer_flushes =
voice_receiver_info.jitter_buffer_flushes;
audio_track_stats->delayed_packet_outage_samples =
@ -1047,6 +1063,7 @@ ProduceMediaStreamTrackStatsFromVoiceReceiverInfo(
audio_track_stats->total_interruption_duration =
static_cast<double>(voice_receiver_info.total_interruption_duration_ms) /
rtc::kNumMillisecsPerSec;
return audio_track_stats;
}

14
pc/rtc_stats_collector_unittest.cc
View file

@ -2234,6 +2234,8 @@ TEST_F(RTCStatsCollectorTest,
voice_receiver_info.silent_concealed_samples = 765;
voice_receiver_info.jitter_buffer_delay_seconds = 3.456;
voice_receiver_info.jitter_buffer_emitted_count = 13;
// TODO(crbug.com/webrtc/14524): These metrics have been moved from "track"
// stats, no need to test these here.
voice_receiver_info.jitter_buffer_flushes = 7;
voice_receiver_info.delayed_packet_outage_samples = 15;
voice_receiver_info.relative_packet_arrival_delay_seconds = 16;
@ -2278,6 +2280,8 @@ TEST_F(RTCStatsCollectorTest,
expected_remote_audio_track.silent_concealed_samples = 765;
expected_remote_audio_track.jitter_buffer_delay = 3.456;
expected_remote_audio_track.jitter_buffer_emitted_count = 13;
// TODO(crbug.com/webrtc/14524): These metrics have been moved from "track"
// stats, delete them.
expected_remote_audio_track.jitter_buffer_flushes = 7;
expected_remote_audio_track.delayed_packet_outage_samples = 15;
expected_remote_audio_track.relative_packet_arrival_delay = 16;
@ -2471,6 +2475,11 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Audio) {
voice_media_info.receivers[0].audio_level = 14442; // [0,32767]
voice_media_info.receivers[0].total_output_energy = 10.0;
voice_media_info.receivers[0].total_output_duration = 11.0;
voice_media_info.receivers[0].jitter_buffer_flushes = 7;
voice_media_info.receivers[0].delayed_packet_outage_samples = 15;
voice_media_info.receivers[0].relative_packet_arrival_delay_seconds = 16;
voice_media_info.receivers[0].interruption_count = 7788;
voice_media_info.receivers[0].total_interruption_duration_ms = 778899;
voice_media_info.receivers[0].last_packet_received_timestamp_ms =
absl::nullopt;
@ -2526,6 +2535,11 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Audio) {
expected_audio.audio_level = 14442.0 / 32767.0; // [0,1]
expected_audio.total_audio_energy = 10.0;
expected_audio.total_samples_duration = 11.0;
expected_audio.jitter_buffer_flushes = 7;
expected_audio.delayed_packet_outage_samples = 15;
expected_audio.relative_packet_arrival_delay = 16;
expected_audio.interruption_count = 7788;
expected_audio.total_interruption_duration = 778.899;
ASSERT_TRUE(report->Get(expected_audio.id()));
EXPECT_EQ(

24
pc/rtc_stats_integrationtest.cc
View file

@ -656,6 +656,8 @@ class RTCStatsReportVerifier {
media_stream_track.inserted_samples_for_deceleration);
verifier.TestMemberIsUndefined(
media_stream_track.removed_samples_for_acceleration);
// TODO(crbug.com/webrtc/14524): These metrics have been moved from
// "track" stats, delete them.
verifier.TestMemberIsUndefined(media_stream_track.jitter_buffer_flushes);
verifier.TestMemberIsUndefined(
media_stream_track.delayed_packet_outage_samples);
@ -691,6 +693,8 @@ class RTCStatsReportVerifier {
media_stream_track.inserted_samples_for_deceleration);
verifier.TestMemberIsNonNegative<uint64_t>(
media_stream_track.removed_samples_for_acceleration);
// TODO(crbug.com/webrtc/14524): These metrics have been moved from
// "track" stats, delete them.
verifier.TestMemberIsNonNegative<uint64_t>(
media_stream_track.jitter_buffer_flushes);
verifier.TestMemberIsNonNegative<uint64_t>(
@ -722,6 +726,8 @@ class RTCStatsReportVerifier {
media_stream_track.inserted_samples_for_deceleration);
verifier.TestMemberIsUndefined(
media_stream_track.removed_samples_for_acceleration);
// TODO(crbug.com/webrtc/14524): These metrics have been moved from
// "track" stats, delete them.
verifier.TestMemberIsUndefined(
media_stream_track.jitter_buffer_flushes);
verifier.TestMemberIsUndefined(
@ -920,6 +926,14 @@ class RTCStatsReportVerifier {
// The integration test is not set up to test screen share; don't require
// this to be present.
verifier.MarkMemberTested(inbound_stream.content_type, true);
verifier.TestMemberIsUndefined(inbound_stream.jitter_buffer_flushes);
verifier.TestMemberIsUndefined(
inbound_stream.delayed_packet_outage_samples);
verifier.TestMemberIsUndefined(
inbound_stream.relative_packet_arrival_delay);
verifier.TestMemberIsUndefined(inbound_stream.interruption_count);
verifier.TestMemberIsUndefined(
inbound_stream.total_interruption_duration);
verifier.TestMemberIsNonNegative<double>(
inbound_stream.min_playout_delay);
} else {
@ -939,6 +953,16 @@ class RTCStatsReportVerifier {
verifier.TestMemberIsUndefined(inbound_stream.freeze_count);
verifier.TestMemberIsUndefined(inbound_stream.total_freezes_duration);
verifier.TestMemberIsUndefined(inbound_stream.content_type);
verifier.TestMemberIsNonNegative<uint64_t>(
inbound_stream.jitter_buffer_flushes);
verifier.TestMemberIsNonNegative<uint64_t>(
inbound_stream.delayed_packet_outage_samples);
verifier.TestMemberIsNonNegative<double>(
inbound_stream.relative_packet_arrival_delay);
verifier.TestMemberIsNonNegative<uint32_t>(
inbound_stream.interruption_count);
verifier.TestMemberIsNonNegative<double>(
inbound_stream.total_interruption_duration);
verifier.TestMemberIsUndefined(inbound_stream.min_playout_delay);
}
return verifier.ExpectAllMembersSuccessfullyTested();

31
pc/test/integration_test_helpers.h
View file

@ -171,6 +171,8 @@ void RemoveSsrcsAndMsids(cricket::SessionDescription* desc);
// endpoint that only signals a=msid lines to convey stream_ids.
void RemoveSsrcsAndKeepMsids(cricket::SessionDescription* desc);
// TODO(https://crbug.com/webrtc/14175): Stop depending on "track" stats, the
// metrics we're interested in are already available in "inbound-rtp".
int FindFirstMediaStatsIndexByKind(
const std::string& kind,
const std::vector<const webrtc::RTCMediaStreamTrackStats*>&
@ -650,34 +652,26 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
void StartWatchingDelayStats() {
// Get the baseline numbers for audio_packets and audio_delay.
auto received_stats = NewGetStats();
auto track_stats =
received_stats->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>()[0];
ASSERT_TRUE(track_stats->relative_packet_arrival_delay.is_defined());
auto rtp_stats =
received_stats->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>()[0];
ASSERT_TRUE(rtp_stats->relative_packet_arrival_delay.is_defined());
ASSERT_TRUE(rtp_stats->packets_received.is_defined());
ASSERT_TRUE(rtp_stats->track_id.is_defined());
audio_track_stats_id_ = track_stats->id();
ASSERT_TRUE(received_stats->Get(audio_track_stats_id_));
rtp_stats_id_ = rtp_stats->id();
ASSERT_EQ(audio_track_stats_id_, *rtp_stats->track_id);
audio_packets_stat_ = *rtp_stats->packets_received;
audio_delay_stat_ = *track_stats->relative_packet_arrival_delay;
audio_samples_stat_ = *track_stats->total_samples_received;
audio_concealed_stat_ = *track_stats->concealed_samples;
audio_delay_stat_ = *rtp_stats->relative_packet_arrival_delay;
audio_samples_stat_ = *rtp_stats->total_samples_received;
audio_concealed_stat_ = *rtp_stats->concealed_samples;
}
void UpdateDelayStats(std::string tag, int desc_size) {
auto report = NewGetStats();
auto track_stats =
report->GetAs<webrtc::RTCMediaStreamTrackStats>(audio_track_stats_id_);
ASSERT_TRUE(track_stats);
auto rtp_stats =
report->GetAs<webrtc::RTCInboundRTPStreamStats>(rtp_stats_id_);
ASSERT_TRUE(rtp_stats);
auto delta_packets = *rtp_stats->packets_received - audio_packets_stat_;
auto delta_rpad =
*track_stats->relative_packet_arrival_delay - audio_delay_stat_;
*rtp_stats->relative_packet_arrival_delay - audio_delay_stat_;
auto recent_delay = delta_packets > 0 ? delta_rpad / delta_packets : -1;
// The purpose of these checks is to sound the alarm early if we introduce
// serious regressions. The numbers are not acceptable for production, but
@ -694,9 +688,9 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
EXPECT_GT(0.1, recent_delay) << tag << " size " << desc_size;
#endif
auto delta_samples =
*track_stats->total_samples_received - audio_samples_stat_;
*rtp_stats->total_samples_received - audio_samples_stat_;
auto delta_concealed =
*track_stats->concealed_samples - audio_concealed_stat_;
*rtp_stats->concealed_samples - audio_concealed_stat_;
// These limits should be adjusted down as we improve:
//
// Concealing more than 4000 samples during a renegotiation is unacceptable.
@ -731,9 +725,9 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
}
// Increment trailing counters
audio_packets_stat_ = *rtp_stats->packets_received;
audio_delay_stat_ = *track_stats->relative_packet_arrival_delay;
audio_samples_stat_ = *track_stats->total_samples_received;
audio_concealed_stat_ = *track_stats->concealed_samples;
audio_delay_stat_ = *rtp_stats->relative_packet_arrival_delay;
audio_samples_stat_ = *rtp_stats->total_samples_received;
audio_concealed_stat_ = *rtp_stats->concealed_samples;
}
// Sets number of candidates expected
@ -1240,7 +1234,6 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
uint64_t audio_samples_stat_ = 0;
uint64_t audio_concealed_stat_ = 0;
std::string rtp_stats_id_;
std::string audio_track_stats_id_;
ScopedTaskSafety task_safety_;

22
stats/rtcstats_objects.cc
View file

@ -696,6 +696,11 @@ WEBRTC_RTCSTATS_IMPL(
&pli_count,
&nack_count,
&qp_sum,
&jitter_buffer_flushes,
&delayed_packet_outage_samples,
&relative_packet_arrival_delay,
&interruption_count,
&total_interruption_duration,
&min_playout_delay)
// clang-format on
@ -754,6 +759,18 @@ RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(std::string&& id,
pli_count("pliCount"),
nack_count("nackCount"),
qp_sum("qpSum"),
jitter_buffer_flushes(
"jitterBufferFlushes",
{NonStandardGroupId::kRtcAudioJitterBufferMaxPackets}),
delayed_packet_outage_samples(
"delayedPacketOutageSamples",
{NonStandardGroupId::kRtcAudioJitterBufferMaxPackets,
NonStandardGroupId::kRtcStatsRelativePacketArrivalDelay}),
relative_packet_arrival_delay(
"relativePacketArrivalDelay",
{NonStandardGroupId::kRtcStatsRelativePacketArrivalDelay}),
interruption_count("interruptionCount"),
total_interruption_duration("totalInterruptionDuration"),
min_playout_delay("minPlayoutDelay") {}
RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(
@ -808,6 +825,11 @@ RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(
pli_count(other.pli_count),
nack_count(other.nack_count),
qp_sum(other.qp_sum),
jitter_buffer_flushes(other.jitter_buffer_flushes),
delayed_packet_outage_samples(other.delayed_packet_outage_samples),
relative_packet_arrival_delay(other.relative_packet_arrival_delay),
interruption_count(other.interruption_count),
total_interruption_duration(other.total_interruption_duration),
min_playout_delay(other.min_playout_delay) {}
RTCInboundRTPStreamStats::~RTCInboundRTPStreamStats() {}