Expose relative packet arrival delay metric in stats API.

The metric is non-standard and documented in: https://github.com/henbos/webrtc-provisional-stats/pull/14

Bug: webrtc:10333
Change-Id: Ie5b4bbad5b1e2c9104742931529bab8f48f51f8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125861
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26999}

api/stats/rtcstats_objects.h

@@ -314,6 +314,7 @@ class RTC_EXPORT RTCMediaStreamTrackStats final : public RTCStats {
   // TODO(kuddai): Add description to standard. crbug.com/webrtc/10042
   RTCNonStandardStatsMember<uint64_t> jitter_buffer_flushes;
   RTCNonStandardStatsMember<uint64_t> delayed_packet_outage_samples;
+  RTCNonStandardStatsMember<double> relative_packet_arrival_delay;
   // Non-standard video-only members.
   // https://henbos.github.io/webrtc-provisional-stats/#RTCVideoReceiverStats-dict*
   RTCNonStandardStatsMember<uint32_t> freeze_count;

audio/audio_receive_stream.cc

@@ -221,6 +221,9 @@ webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
   stats.preemptive_expand_rate = Q14ToFloat(ns.currentPreemptiveRate);
   stats.jitter_buffer_flushes = ns.packetBufferFlushes;
   stats.delayed_packet_outage_samples = ns.delayedPacketOutageSamples;
+  stats.relative_packet_arrival_delay_seconds =
+      static_cast<double>(ns.relativePacketArrivalDelayMs) /
+      static_cast<double>(rtc::kNumMillisecsPerSec);
 
   auto ds = channel_receive_->GetDecodingCallStatistics();
   stats.decoding_calls_to_silence_generator = ds.calls_to_silence_generator;

call/audio_receive_stream.h

@@ -74,6 +74,7 @@ class AudioReceiveStream {
     int32_t decoding_muted_output = 0;
     int64_t capture_start_ntp_time_ms = 0;
     uint64_t jitter_buffer_flushes = 0;
+    double relative_packet_arrival_delay_seconds = 0.0;
   };
 
   struct Config {

media/base/media_channel.h

@@ -487,6 +487,8 @@ struct VoiceReceiverInfo : public MediaReceiverInfo {
   uint64_t jitter_buffer_flushes = 0;
   // Number of samples expanded due to delayed packets.
   uint64_t delayed_packet_outage_samples = 0;
+  // Arrival delay of received audio packets.
+  double relative_packet_arrival_delay_seconds = 0.0;
 };
 
 struct VideoSenderInfo : public MediaSenderInfo {

media/engine/webrtc_voice_engine.cc

@@ -2278,6 +2278,8 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
     rinfo.decoding_muted_output = stats.decoding_muted_output;
     rinfo.capture_start_ntp_time_ms = stats.capture_start_ntp_time_ms;
     rinfo.jitter_buffer_flushes = stats.jitter_buffer_flushes;
+    rinfo.relative_packet_arrival_delay_seconds =
+        stats.relative_packet_arrival_delay_seconds;
 
     info->receivers.push_back(rinfo);
   }

modules/audio_coding/acm2/acm_receiver.cc

@@ -257,6 +257,8 @@ void AcmReceiver::GetNetworkStatistics(NetworkStatistics* acm_stat) {
       neteq_lifetime_stat.jitter_buffer_emitted_count;
   acm_stat->delayedPacketOutageSamples =
       neteq_lifetime_stat.delayed_packet_outage_samples;
+  acm_stat->relativePacketArrivalDelayMs =
+      neteq_lifetime_stat.relative_packet_arrival_delay_ms;
 
   NetEqOperationsAndState neteq_operations_and_state =
       neteq_->GetOperationsAndState();

modules/audio_coding/include/audio_coding_module_typedefs.h

@@ -122,6 +122,8 @@ struct NetworkStatistics {
   uint64_t packetBufferFlushes;
   // number of samples expanded due to delayed packets
   uint64_t delayedPacketOutageSamples;
+  // arrival delay of incoming packets
+  uint64_t relativePacketArrivalDelayMs;
 };
 
 }  // namespace webrtc

pc/rtc_stats_collector.cc

@@ -460,6 +460,8 @@ ProduceMediaStreamTrackStatsFromVoiceReceiverInfo(
       voice_receiver_info.jitter_buffer_flushes;
   audio_track_stats->delayed_packet_outage_samples =
       voice_receiver_info.delayed_packet_outage_samples;
+  audio_track_stats->relative_packet_arrival_delay =
+      voice_receiver_info.relative_packet_arrival_delay_seconds;
   return audio_track_stats;
 }
 

pc/rtc_stats_collector_unittest.cc

@@ -1428,6 +1428,7 @@ TEST_F(RTCStatsCollectorTest,
   voice_receiver_info.jitter_buffer_emitted_count = 13;
   voice_receiver_info.jitter_buffer_flushes = 7;
   voice_receiver_info.delayed_packet_outage_samples = 15;
+  voice_receiver_info.relative_packet_arrival_delay_seconds = 16;
 
   stats_->CreateMockRtpSendersReceiversAndChannels(
       {}, {std::make_pair(remote_audio_track.get(), voice_receiver_info)}, {},
@@ -1464,6 +1465,7 @@ TEST_F(RTCStatsCollectorTest,
   expected_remote_audio_track.jitter_buffer_emitted_count = 13;
   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;
   ASSERT_TRUE(report->Get(expected_remote_audio_track.id()));
   EXPECT_EQ(expected_remote_audio_track,
             report->Get(expected_remote_audio_track.id())

pc/rtc_stats_integrationtest.cc

@@ -652,6 +652,8 @@ class RTCStatsReportVerifier {
           media_stream_track.jitter_buffer_flushes);
       verifier.TestMemberIsNonNegative<uint64_t>(
           media_stream_track.delayed_packet_outage_samples);
+      verifier.TestMemberIsNonNegative<double>(
+          media_stream_track.relative_packet_arrival_delay);
     } else {
       verifier.TestMemberIsUndefined(media_stream_track.jitter_buffer_delay);
       verifier.TestMemberIsUndefined(
@@ -662,6 +664,8 @@ class RTCStatsReportVerifier {
       verifier.TestMemberIsUndefined(media_stream_track.jitter_buffer_flushes);
       verifier.TestMemberIsUndefined(
           media_stream_track.delayed_packet_outage_samples);
+      verifier.TestMemberIsUndefined(
+          media_stream_track.relative_packet_arrival_delay);
     }
     return verifier.ExpectAllMembersSuccessfullyTested();
   }

stats/rtcstats_objects.cc

@@ -382,6 +382,7 @@ WEBRTC_RTCSTATS_IMPL(RTCMediaStreamTrackStats, RTCStats, "track",
                      &concealment_events,
                      &jitter_buffer_flushes,
                      &delayed_packet_outage_samples,
+                     &relative_packet_arrival_delay,
                      &freeze_count,
                      &pause_count,
                      &total_freezes_duration,
@@ -427,6 +428,7 @@ RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(std::string&& id,
       concealment_events("concealmentEvents"),
       jitter_buffer_flushes("jitterBufferFlushes"),
       delayed_packet_outage_samples("delayedPacketOutageSamples"),
+      relative_packet_arrival_delay("relativePacketArrivalDelay"),
       freeze_count("freezeCount"),
       pause_count("pauseCount"),
       total_freezes_duration("totalFreezesDuration"),
@@ -468,6 +470,7 @@ RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(
       concealment_events(other.concealment_events),
       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),
       freeze_count(other.freeze_count),
       pause_count(other.pause_count),
       total_freezes_duration(other.total_freezes_duration),