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Expose new audio stats on the API

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Several new audio stats were recently standardized and implemented in
WebRTC in https://webrtc-review.googlesource.com/c/src/+/133887. This CL
adds these to the GetStats API.

Bug: webrtc:10442, webrtc:10443, webrtc:10444
Change-Id: I0e898ac14777e82b1a9099b5e0a5584eb9cb5934
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134213
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27839}
This commit is contained in:
Ivo Creusen 2019-04-30 09:45:21 +02:00 committed by Commit Bot
parent e847481dc8
commit 8d8ffdbcca
12 changed files with 88 additions and 3 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

5
api/stats/rtcstats_objects.h
View File

@ -315,7 +315,10 @@ class RTC_EXPORT RTCMediaStreamTrackStats final : public RTCStats {
RTCStatsMember<uint64_t> total_samples_received;
RTCStatsMember<double> total_samples_duration;
RTCStatsMember<uint64_t> concealed_samples;
RTCStatsMember<uint64_t> silent_concealed_samples;
RTCStatsMember<uint64_t> concealment_events;
RTCStatsMember<uint64_t> inserted_samples_for_deceleration;
RTCStatsMember<uint64_t> removed_samples_for_acceleration;
// Non-standard audio-only member
// TODO(kuddai): Add description to standard. crbug.com/webrtc/10042
RTCNonStandardStatsMember<uint64_t> jitter_buffer_flushes;
@ -399,6 +402,8 @@ class RTC_EXPORT RTCInboundRTPStreamStats final : public RTCRTPStreamStats {
~RTCInboundRTPStreamStats() override;
RTCStatsMember<uint32_t> packets_received;
RTCStatsMember<uint64_t> fec_packets_received;
RTCStatsMember<uint64_t> fec_packets_discarded;
RTCStatsMember<uint64_t> bytes_received;
RTCStatsMember<int32_t> packets_lost; // Signed per RFC 3550
RTCStatsMember<double> last_packet_received_timestamp;

5
audio/audio_receive_stream.cc
View File

@ -206,15 +206,20 @@ webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
// Get jitter buffer and total delay (alg + jitter + playout) stats.
auto ns = channel_receive_->GetNetworkStatistics();
stats.fec_packets_received = ns.fecPacketsReceived;
stats.fec_packets_discarded = ns.fecPacketsDiscarded;
stats.jitter_buffer_ms = ns.currentBufferSize;
stats.jitter_buffer_preferred_ms = ns.preferredBufferSize;
stats.total_samples_received = ns.totalSamplesReceived;
stats.concealed_samples = ns.concealedSamples;
stats.silent_concealed_samples = ns.silentConcealedSamples;
stats.concealment_events = ns.concealmentEvents;
stats.jitter_buffer_delay_seconds =
static_cast<double>(ns.jitterBufferDelayMs) /
static_cast<double>(rtc::kNumMillisecsPerSec);
stats.jitter_buffer_emitted_count = ns.jitterBufferEmittedCount;
stats.inserted_samples_for_deceleration = ns.insertedSamplesForDeceleration;
stats.removed_samples_for_acceleration = ns.removedSamplesForAcceleration;
stats.expand_rate = Q14ToFloat(ns.currentExpandRate);
stats.speech_expand_rate = Q14ToFloat(ns.currentSpeechExpandRate);
stats.secondary_decoded_rate = Q14ToFloat(ns.currentSecondaryDecodedRate);

5
audio/audio_receive_stream_unittest.cc
View File

@ -67,8 +67,9 @@ const std::pair<int, SdpAudioFormat> kReceiveCodec = {
123,
{"codec_name_recv", 96000, 0}};
const NetworkStatistics kNetworkStats = {
123, 456, false, 789012, 3456, 123, 456, 789, 0, {}, 789,
12, 345, 678, 901, 0, -1, -1, -1, -1, -1, 0};
123, 456, false, 789012, 3456, 123, 456, 789, 543, 432,
321, 123, 101, 0, {}, 789, 12, 345, 678, 901,
0, -1, -1, -1, -1, -1, 0, 0, 0, 0};
const AudioDecodingCallStats kAudioDecodeStats = MakeAudioDecodeStatsForTest();
struct ConfigHelper {

5
call/audio_receive_stream.h
View File

@ -38,6 +38,8 @@ class AudioReceiveStream {
uint32_t remote_ssrc = 0;
int64_t bytes_rcvd = 0;
uint32_t packets_rcvd = 0;
uint64_t fec_packets_received = 0;
uint64_t fec_packets_discarded = 0;
uint32_t packets_lost = 0;
float fraction_lost = 0.0f;
std::string codec_name;
@ -54,9 +56,12 @@ class AudioReceiveStream {
uint64_t total_samples_received = 0;
double total_output_duration = 0.0;
uint64_t concealed_samples = 0;
uint64_t silent_concealed_samples = 0;
uint64_t concealment_events = 0;
double jitter_buffer_delay_seconds = 0.0;
uint64_t jitter_buffer_emitted_count = 0;
uint64_t inserted_samples_for_deceleration = 0;
uint64_t removed_samples_for_acceleration = 0;
// Stats below DO NOT correspond directly to anything in the WebRTC stats
float expand_rate = 0.0f;
float speech_expand_rate = 0.0f;

5
media/base/media_channel.h
View File

@ -479,9 +479,14 @@ struct VoiceReceiverInfo : public MediaReceiverInfo {
uint64_t total_samples_received = 0;
double total_output_duration = 0.0;
uint64_t concealed_samples = 0;
uint64_t silent_concealed_samples = 0;
uint64_t concealment_events = 0;
double jitter_buffer_delay_seconds = 0.0;
uint64_t jitter_buffer_emitted_count = 0;
uint64_t inserted_samples_for_deceleration = 0;
uint64_t removed_samples_for_acceleration = 0;
uint64_t fec_packets_received = 0;
uint64_t fec_packets_discarded = 0;
// Stats below DO NOT correspond directly to anything in the WebRTC stats
// fraction of synthesized audio inserted through expansion.
float expand_rate = 0.0f;

7
media/engine/webrtc_voice_engine.cc
View File

@ -2226,6 +2226,8 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
rinfo.add_ssrc(stats.remote_ssrc);
rinfo.bytes_rcvd = stats.bytes_rcvd;
rinfo.packets_rcvd = stats.packets_rcvd;
rinfo.fec_packets_received = stats.fec_packets_received;
rinfo.fec_packets_discarded = stats.fec_packets_discarded;
rinfo.packets_lost = stats.packets_lost;
rinfo.fraction_lost = stats.fraction_lost;
rinfo.codec_name = stats.codec_name;
@ -2240,9 +2242,14 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
rinfo.total_samples_received = stats.total_samples_received;
rinfo.total_output_duration = stats.total_output_duration;
rinfo.concealed_samples = stats.concealed_samples;
rinfo.silent_concealed_samples = stats.silent_concealed_samples;
rinfo.concealment_events = stats.concealment_events;
rinfo.jitter_buffer_delay_seconds = stats.jitter_buffer_delay_seconds;
rinfo.jitter_buffer_emitted_count = stats.jitter_buffer_emitted_count;
rinfo.inserted_samples_for_deceleration =
stats.inserted_samples_for_deceleration;
rinfo.removed_samples_for_acceleration =
stats.removed_samples_for_acceleration;
rinfo.expand_rate = stats.expand_rate;
rinfo.speech_expand_rate = stats.speech_expand_rate;
rinfo.secondary_decoded_rate = stats.secondary_decoded_rate;

8
modules/audio_coding/acm2/acm_receiver.cc
View File

@ -251,6 +251,8 @@ void AcmReceiver::GetNetworkStatistics(NetworkStatistics* acm_stat) {
NetEqLifetimeStatistics neteq_lifetime_stat = neteq_->GetLifetimeStatistics();
acm_stat->totalSamplesReceived = neteq_lifetime_stat.total_samples_received;
acm_stat->concealedSamples = neteq_lifetime_stat.concealed_samples;
acm_stat->silentConcealedSamples =
neteq_lifetime_stat.silent_concealed_samples;
acm_stat->concealmentEvents = neteq_lifetime_stat.concealment_events;
acm_stat->jitterBufferDelayMs = neteq_lifetime_stat.jitter_buffer_delay_ms;
acm_stat->jitterBufferEmittedCount =
@ -262,6 +264,12 @@ void AcmReceiver::GetNetworkStatistics(NetworkStatistics* acm_stat) {
acm_stat->interruptionCount = neteq_lifetime_stat.interruption_count;
acm_stat->totalInterruptionDurationMs =
neteq_lifetime_stat.total_interruption_duration_ms;
acm_stat->insertedSamplesForDeceleration =
neteq_lifetime_stat.inserted_samples_for_deceleration;
acm_stat->removedSamplesForAcceleration =
neteq_lifetime_stat.removed_samples_for_acceleration;
acm_stat->fecPacketsReceived = neteq_lifetime_stat.fec_packets_received;
acm_stat->fecPacketsDiscarded = neteq_lifetime_stat.fec_packets_discarded;
NetEqOperationsAndState neteq_operations_and_state =
neteq_->GetOperationsAndState();

5
modules/audio_coding/include/audio_coding_module_typedefs.h
View File

@ -84,9 +84,14 @@ struct NetworkStatistics {
// https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats
uint64_t totalSamplesReceived;
uint64_t concealedSamples;
uint64_t silentConcealedSamples;
uint64_t concealmentEvents;
uint64_t jitterBufferDelayMs;
uint64_t jitterBufferEmittedCount;
uint64_t insertedSamplesForDeceleration;
uint64_t removedSamplesForAcceleration;
uint64_t fecPacketsReceived;
uint64_t fecPacketsDiscarded;
// Stats below DO NOT correspond directly to anything in the WebRTC stats
// Loss rate (network + late); fraction between 0 and 1, scaled to Q14.
uint16_t currentPacketLossRate;

12
pc/rtc_stats_collector.cc
View File

@ -250,6 +250,10 @@ void SetInboundRTPStreamStatsFromVoiceReceiverInfo(
*voice_receiver_info.last_packet_received_timestamp_ms) /
rtc::kNumMillisecsPerSec;
}
inbound_audio->fec_packets_received =
voice_receiver_info.fec_packets_received;
inbound_audio->fec_packets_discarded =
voice_receiver_info.fec_packets_discarded;
}
void SetInboundRTPStreamStatsFromVideoReceiverInfo(
@ -475,6 +479,10 @@ ProduceMediaStreamTrackStatsFromVoiceReceiverInfo(
voice_receiver_info.jitter_buffer_delay_seconds;
audio_track_stats->jitter_buffer_emitted_count =
voice_receiver_info.jitter_buffer_emitted_count;
audio_track_stats->inserted_samples_for_deceleration =
voice_receiver_info.inserted_samples_for_deceleration;
audio_track_stats->removed_samples_for_acceleration =
voice_receiver_info.removed_samples_for_acceleration;
audio_track_stats->total_audio_energy =
voice_receiver_info.total_output_energy;
audio_track_stats->total_samples_received =
@ -482,6 +490,8 @@ ProduceMediaStreamTrackStatsFromVoiceReceiverInfo(
audio_track_stats->total_samples_duration =
voice_receiver_info.total_output_duration;
audio_track_stats->concealed_samples = voice_receiver_info.concealed_samples;
audio_track_stats->silent_concealed_samples =
voice_receiver_info.silent_concealed_samples;
audio_track_stats->concealment_events =
voice_receiver_info.concealment_events;
audio_track_stats->jitter_buffer_flushes =
@ -921,7 +931,7 @@ void RTCStatsCollector::ProducePartialResultsOnSignalingThreadImpl(
void RTCStatsCollector::ProducePartialResultsOnNetworkThread(
int64_t timestamp_us) {
RTC_DCHECK(network_thread_->IsCurrent());
// Touching |network_report_| on this thread is safe by this method because
// Touching |network_report_| on this thread is safe by this method because
// |network_report_event_| is reset before this method is invoked.
network_report_ = RTCStatsReport::Create(timestamp_us);

10
pc/rtc_stats_collector_unittest.cc
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@ -1424,6 +1424,9 @@ TEST_F(RTCStatsCollectorTest,
voice_receiver_info.total_output_duration = 0.25;
voice_receiver_info.concealed_samples = 123;
voice_receiver_info.concealment_events = 12;
voice_receiver_info.inserted_samples_for_deceleration = 987;
voice_receiver_info.removed_samples_for_acceleration = 876;
voice_receiver_info.silent_concealed_samples = 765;
voice_receiver_info.jitter_buffer_delay_seconds = 3456;
voice_receiver_info.jitter_buffer_emitted_count = 13;
voice_receiver_info.jitter_buffer_flushes = 7;
@ -1463,6 +1466,9 @@ TEST_F(RTCStatsCollectorTest,
expected_remote_audio_track.total_samples_duration = 0.25;
expected_remote_audio_track.concealed_samples = 123;
expected_remote_audio_track.concealment_events = 12;
expected_remote_audio_track.inserted_samples_for_deceleration = 987;
expected_remote_audio_track.removed_samples_for_acceleration = 876;
expected_remote_audio_track.silent_concealed_samples = 765;
expected_remote_audio_track.jitter_buffer_delay = 3456;
expected_remote_audio_track.jitter_buffer_emitted_count = 13;
expected_remote_audio_track.jitter_buffer_flushes = 7;
@ -1625,6 +1631,8 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Audio) {
voice_media_info.receivers[0].local_stats[0].ssrc = 1;
voice_media_info.receivers[0].packets_lost = -1; // Signed per RFC3550
voice_media_info.receivers[0].packets_rcvd = 2;
voice_media_info.receivers[0].fec_packets_discarded = 5566;
voice_media_info.receivers[0].fec_packets_received = 6677;
voice_media_info.receivers[0].bytes_rcvd = 3;
voice_media_info.receivers[0].codec_payload_type = 42;
voice_media_info.receivers[0].jitter_ms = 4500;
@ -1660,6 +1668,8 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Audio) {
expected_audio.transport_id = "RTCTransport_TransportName_1";
expected_audio.codec_id = "RTCCodec_AudioMid_Inbound_42";
expected_audio.packets_received = 2;
expected_audio.fec_packets_discarded = 5566;
expected_audio.fec_packets_received = 6677;
expected_audio.bytes_received = 3;
expected_audio.packets_lost = -1;
// |expected_audio.last_packet_received_timestamp| should be undefined.

13
pc/rtc_stats_integrationtest.cc
View File

@ -648,6 +648,12 @@ class RTCStatsReportVerifier {
media_stream_track.concealed_samples);
verifier.TestMemberIsNonNegative<uint64_t>(
media_stream_track.concealment_events);
verifier.TestMemberIsNonNegative<uint64_t>(
media_stream_track.inserted_samples_for_deceleration);
verifier.TestMemberIsNonNegative<uint64_t>(
media_stream_track.removed_samples_for_acceleration);
verifier.TestMemberIsNonNegative<uint64_t>(
media_stream_track.silent_concealed_samples);
verifier.TestMemberIsNonNegative<uint64_t>(
media_stream_track.jitter_buffer_flushes);
verifier.TestMemberIsNonNegative<uint64_t>(
@ -722,6 +728,13 @@ class RTCStatsReportVerifier {
verifier.TestMemberIsUndefined(inbound_stream.qp_sum);
}
verifier.TestMemberIsNonNegative<uint32_t>(inbound_stream.packets_received);
if (inbound_stream.media_type.is_defined() &&
*inbound_stream.media_type == "audio") {
verifier.TestMemberIsNonNegative<uint64_t>(
inbound_stream.fec_packets_received);
verifier.TestMemberIsNonNegative<uint64_t>(
inbound_stream.fec_packets_discarded);
}
verifier.TestMemberIsNonNegative<uint64_t>(inbound_stream.bytes_received);
// packets_lost is defined as signed, but this should never happen in
// this test. See RFC 3550.

11
stats/rtcstats_objects.cc
View File

@ -433,7 +433,10 @@ RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(std::string&& id,
total_samples_received("totalSamplesReceived"),
total_samples_duration("totalSamplesDuration"),
concealed_samples("concealedSamples"),
silent_concealed_samples("silentConcealedSamples"),
concealment_events("concealmentEvents"),
inserted_samples_for_deceleration("insertedSamplesForDeceleration"),
removed_samples_for_acceleration("removedSamplesForAcceleration"),
jitter_buffer_flushes(
"jitterBufferFlushes",
{NonStandardGroupId::kRtcAudioJitterBufferMaxPackets}),
@ -484,7 +487,11 @@ RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(
total_samples_received(other.total_samples_received),
total_samples_duration(other.total_samples_duration),
concealed_samples(other.concealed_samples),
silent_concealed_samples(other.silent_concealed_samples),
concealment_events(other.concealment_events),
inserted_samples_for_deceleration(
other.inserted_samples_for_deceleration),
removed_samples_for_acceleration(other.removed_samples_for_acceleration),
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),
@ -610,6 +617,8 @@ RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(std::string&& id,
int64_t timestamp_us)
: RTCRTPStreamStats(std::move(id), timestamp_us),
packets_received("packetsReceived"),
fec_packets_received("fecPacketsReceived"),
fec_packets_discarded("fecPacketsDiscarded"),
bytes_received("bytesReceived"),
packets_lost("packetsLost"),
last_packet_received_timestamp("lastPacketReceivedTimestamp"),
@ -633,6 +642,8 @@ RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(
const RTCInboundRTPStreamStats& other)
: RTCRTPStreamStats(other),
packets_received(other.packets_received),
fec_packets_received(other.fec_packets_received),
fec_packets_discarded(other.fec_packets_discarded),
bytes_received(other.bytes_received),
packets_lost(other.packets_lost),
last_packet_received_timestamp(other.last_packet_received_timestamp),