stats: implement remote-outbound-rtp for video

following the audio changes. Note that RTT-related fields require
DLRR and are not implemented yet.

BUG=webrtc:12529

Change-Id: I3f9449fbe876a1b282a32f2bcebe1cf3e10989bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346580
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42069}

call/video_receive_stream.h

@@ -115,9 +115,11 @@ class VideoReceiveStreamInterface : public MediaReceiveStreamInterface {
     TimeDelta total_decode_time = TimeDelta::Zero();
     // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay
     TimeDelta total_processing_delay = TimeDelta::Zero();
-    // TODO(bugs.webrtc.org/13986): standardize
+
+    // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalassemblytime
     TimeDelta total_assembly_time = TimeDelta::Zero();
     uint32_t frames_assembled_from_multiple_packets = 0;
+
     // Total inter frame delay in seconds.
     // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalinterframedelay
     double total_inter_frame_delay = 0;
@@ -154,6 +156,14 @@ class VideoReceiveStreamInterface : public MediaReceiveStreamInterface {
     // Timing frame info: all important timestamps for a full lifetime of a
     // single 'timing frame'.
     absl::optional<webrtc::TimingFrameInfo> timing_frame_info;
+
+    // Remote outbound stats derived by the received RTCP sender reports.
+    // https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict*
+    absl::optional<int64_t> last_sender_report_timestamp_ms;
+    absl::optional<int64_t> last_sender_report_remote_timestamp_ms;
+    uint32_t sender_reports_packets_sent = 0;
+    uint64_t sender_reports_bytes_sent = 0;
+    uint64_t sender_reports_reports_count = 0;
   };
 
   struct Config {

media/base/media_channel.h

@@ -476,6 +476,19 @@ struct MediaReceiverInfo {
   absl::optional<uint64_t> fec_packets_discarded;
   // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-fecbytesreceived
   absl::optional<uint64_t> fec_bytes_received;
+
+  // Remote outbound stats derived by the received RTCP sender reports.
+  // https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict*
+  absl::optional<int64_t> last_sender_report_timestamp_ms;
+  absl::optional<int64_t> last_sender_report_remote_timestamp_ms;
+  uint64_t sender_reports_packets_sent = 0;
+  uint64_t sender_reports_bytes_sent = 0;
+  uint64_t sender_reports_reports_count = 0;
+  // These require a DLRR block, see
+  // https://w3c.github.io/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
+  absl::optional<webrtc::TimeDelta> round_trip_time;
+  webrtc::TimeDelta total_round_trip_time = webrtc::TimeDelta::Zero();
+  int round_trip_time_measurements = 0;
 };
 
 struct VoiceSenderInfo : public MediaSenderInfo {
@@ -551,16 +564,6 @@ struct VoiceReceiverInfo : public MediaReceiverInfo {
   // longer than 150 ms).
   int32_t interruption_count = 0;
   int32_t total_interruption_duration_ms = 0;
-  // Remote outbound stats derived by the received RTCP sender reports.
-  // https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict*
-  absl::optional<int64_t> last_sender_report_timestamp_ms;
-  absl::optional<int64_t> last_sender_report_remote_timestamp_ms;
-  uint64_t sender_reports_packets_sent = 0;
-  uint64_t sender_reports_bytes_sent = 0;
-  uint64_t sender_reports_reports_count = 0;
-  absl::optional<webrtc::TimeDelta> round_trip_time;
-  webrtc::TimeDelta total_round_trip_time = webrtc::TimeDelta::Zero();
-  int round_trip_time_measurements = 0;
 };
 
 struct VideoSenderInfo : public MediaSenderInfo {

media/engine/webrtc_video_engine.cc

@@ -3737,6 +3737,16 @@ WebRtcVideoReceiveChannel::WebRtcVideoReceiveStream::GetVideoReceiverInfo(
     }
   }
 
+  // remote-outbound-rtp stats.
+  info.last_sender_report_timestamp_ms = stats.last_sender_report_timestamp_ms;
+  info.last_sender_report_remote_timestamp_ms =
+      stats.last_sender_report_remote_timestamp_ms;
+  info.sender_reports_packets_sent = stats.sender_reports_packets_sent;
+  info.sender_reports_bytes_sent = stats.sender_reports_bytes_sent;
+  info.sender_reports_reports_count = stats.sender_reports_reports_count;
+  // TODO(bugs.webrtc.org/12529): RTT-related fields are missing and can only be
+  // present if DLRR is enabled.
+
   if (log_stats)
     RTC_LOG(LS_INFO) << stats.ToString(rtc::TimeMillis());
 

pc/rtc_stats_collector.cc

@@ -514,53 +514,54 @@ std::unique_ptr<RTCAudioPlayoutStats> CreateAudioPlayoutStats(
 }
 
 std::unique_ptr<RTCRemoteOutboundRtpStreamStats>
-CreateRemoteOutboundAudioStreamStats(
-    const cricket::VoiceReceiverInfo& voice_receiver_info,
+CreateRemoteOutboundMediaStreamStats(
+    const cricket::MediaReceiverInfo& media_receiver_info,
     const std::string& mid,
+    const std::string& kind,
     const RTCInboundRtpStreamStats& inbound_audio_stats,
     const std::string& transport_id) {
-  if (!voice_receiver_info.last_sender_report_timestamp_ms.has_value()) {
+  if (!media_receiver_info.last_sender_report_timestamp_ms.has_value()) {
     // Cannot create `RTCRemoteOutboundRtpStreamStats` when the RTCP SR arrival
     // timestamp is not available - i.e., until the first sender report is
     // received.
     return nullptr;
   }
-  RTC_DCHECK_GT(voice_receiver_info.sender_reports_reports_count, 0);
+  RTC_DCHECK_GT(media_receiver_info.sender_reports_reports_count, 0);
 
   // Create.
   auto stats = std::make_unique<RTCRemoteOutboundRtpStreamStats>(
       /*id=*/RTCRemoteOutboundRTPStreamStatsIDFromSSRC(
-          cricket::MEDIA_TYPE_AUDIO, voice_receiver_info.ssrc()),
-      Timestamp::Millis(*voice_receiver_info.last_sender_report_timestamp_ms));
+          cricket::MEDIA_TYPE_AUDIO, media_receiver_info.ssrc()),
+      Timestamp::Millis(*media_receiver_info.last_sender_report_timestamp_ms));
 
   // Populate.
   // - RTCRtpStreamStats.
-  stats->ssrc = voice_receiver_info.ssrc();
-  stats->kind = "audio";
+  stats->ssrc = media_receiver_info.ssrc();
+  stats->kind = kind;
   stats->transport_id = transport_id;
   if (inbound_audio_stats.codec_id.has_value()) {
     stats->codec_id = *inbound_audio_stats.codec_id;
   }
   // - RTCSentRtpStreamStats.
-  stats->packets_sent = voice_receiver_info.sender_reports_packets_sent;
-  stats->bytes_sent = voice_receiver_info.sender_reports_bytes_sent;
+  stats->packets_sent = media_receiver_info.sender_reports_packets_sent;
+  stats->bytes_sent = media_receiver_info.sender_reports_bytes_sent;
   // - RTCRemoteOutboundRtpStreamStats.
   stats->local_id = inbound_audio_stats.id();
   // last_sender_report_remote_timestamp_ms is set together with
   // last_sender_report_timestamp_ms.
   RTC_DCHECK(
-      voice_receiver_info.last_sender_report_remote_timestamp_ms.has_value());
+      media_receiver_info.last_sender_report_remote_timestamp_ms.has_value());
   stats->remote_timestamp = static_cast<double>(
-      *voice_receiver_info.last_sender_report_remote_timestamp_ms);
-  stats->reports_sent = voice_receiver_info.sender_reports_reports_count;
-  if (voice_receiver_info.round_trip_time.has_value()) {
+      *media_receiver_info.last_sender_report_remote_timestamp_ms);
+  stats->reports_sent = media_receiver_info.sender_reports_reports_count;
+  if (media_receiver_info.round_trip_time.has_value()) {
     stats->round_trip_time =
-        voice_receiver_info.round_trip_time->seconds<double>();
+        media_receiver_info.round_trip_time->seconds<double>();
   }
   stats->round_trip_time_measurements =
-      voice_receiver_info.round_trip_time_measurements;
+      media_receiver_info.round_trip_time_measurements;
   stats->total_round_trip_time =
-      voice_receiver_info.total_round_trip_time.seconds<double>();
+      media_receiver_info.total_round_trip_time.seconds<double>();
 
   return stats;
 }
@@ -1709,8 +1710,8 @@ void RTCStatsCollector::ProduceAudioRTPStreamStats_n(
       continue;
     }
     // Remote-outbound.
-    auto remote_outbound_audio = CreateRemoteOutboundAudioStreamStats(
-        voice_receiver_info, mid, *inbound_audio_ptr, transport_id);
+    auto remote_outbound_audio = CreateRemoteOutboundMediaStreamStats(
+        voice_receiver_info, mid, "audio", *inbound_audio_ptr, transport_id);
     // Add stats.
     if (remote_outbound_audio) {
       // When the remote outbound stats are available, the remote ID for the
@@ -1782,7 +1783,7 @@ void RTCStatsCollector::ProduceVideoRTPStreamStats_n(
   std::string mid = *stats.mid;
   std::string transport_id = RTCTransportStatsIDFromTransportChannel(
       *stats.transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
-  // Inbound
+  // Inbound and remote-outbound.
   for (const cricket::VideoReceiverInfo& video_receiver_info :
        stats.track_media_info_map.video_media_info()->receivers) {
     if (!video_receiver_info.connected())
@@ -1795,9 +1796,27 @@ void RTCStatsCollector::ProduceVideoRTPStreamStats_n(
     if (video_track) {
       inbound_video->track_identifier = video_track->id();
     }
-    if (!report->TryAddStats(std::move(inbound_video))) {
+    auto* inbound_video_ptr = report->TryAddStats(std::move(inbound_video));
+    if (!inbound_video_ptr) {
       RTC_LOG(LS_ERROR)
           << "Unable to add video 'inbound-rtp' to report, ID is not unique.";
+      continue;
+    }
+    // Remote-outbound.
+    auto remote_outbound_video = CreateRemoteOutboundMediaStreamStats(
+        video_receiver_info, mid, "video", *inbound_video_ptr, transport_id);
+    // Add stats.
+    if (remote_outbound_video) {
+      // When the remote outbound stats are available, the remote ID for the
+      // local inbound stats is set.
+      auto* remote_outbound_video_ptr =
+          report->TryAddStats(std::move(remote_outbound_video));
+      if (remote_outbound_video_ptr) {
+        inbound_video_ptr->remote_id = remote_outbound_video_ptr->id();
+      } else {
+        RTC_LOG(LS_ERROR) << "Unable to add video 'remote-outbound-rtp' to "
+                          << "report, ID is not unique.";
+      }
     }
   }
   // Outbound

pc/rtc_stats_integrationtest.cc

@@ -939,10 +939,16 @@ class RTCStatsReportVerifier {
     VerifyRTCRtpStreamStats(remote_outbound_stream, verifier);
     VerifyRTCSentRtpStreamStats(remote_outbound_stream, verifier);
     verifier.TestAttributeIsIDReference(remote_outbound_stream.local_id,
-                                        RTCOutboundRtpStreamStats::kType);
+                                        RTCInboundRtpStreamStats::kType);
     verifier.TestAttributeIsNonNegative<double>(
         remote_outbound_stream.remote_timestamp);
     verifier.TestAttributeIsDefined(remote_outbound_stream.reports_sent);
+    // RTT-related attributes need DLRR.
+    verifier.MarkAttributeTested(remote_outbound_stream.round_trip_time, true);
+    verifier.MarkAttributeTested(
+        remote_outbound_stream.round_trip_time_measurements, true);
+    verifier.MarkAttributeTested(remote_outbound_stream.total_round_trip_time,
+                                 true);
     return verifier.ExpectAllAttributesSuccessfullyTested();
   }
 

video/rtp_video_stream_receiver2.cc

@@ -1048,6 +1048,12 @@ absl::optional<int64_t> RtpVideoStreamReceiver2::LastReceivedKeyframePacketMs()
   return absl::nullopt;
 }
 
+absl::optional<RtpRtcpInterface::SenderReportStats>
+RtpVideoStreamReceiver2::GetSenderReportStats() const {
+  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
+  return rtp_rtcp_->GetSenderReportStats();
+}
+
 void RtpVideoStreamReceiver2::ManageFrame(
     std::unique_ptr<RtpFrameObject> frame) {
   RTC_DCHECK_RUN_ON(&packet_sequence_checker_);

video/rtp_video_stream_receiver2.h

@@ -209,6 +209,9 @@ class RtpVideoStreamReceiver2 : public LossNotificationSender,
   absl::optional<uint32_t> LastReceivedFrameRtpTimestamp() const;
   absl::optional<int64_t> LastReceivedKeyframePacketMs() const;
 
+  absl::optional<RtpRtcpInterface::SenderReportStats> GetSenderReportStats()
+      const;
+
  private:
   // Implements RtpVideoFrameReceiver.
   void ManageFrame(std::unique_ptr<RtpFrameObject> frame) override;

video/video_receive_stream2.cc

@@ -568,6 +568,18 @@ VideoReceiveStreamInterface::Stats VideoReceiveStream2::GetStats() const {
       stats.rtx_rtp_stats = rtx_statistician->GetStats();
     }
   }
+
+  absl::optional<RtpRtcpInterface::SenderReportStats> rtcp_sr_stats =
+      rtp_video_stream_receiver_.GetSenderReportStats();
+  if (rtcp_sr_stats) {
+    stats.last_sender_report_timestamp_ms =
+        rtcp_sr_stats->last_arrival_timestamp.ToMs();
+    stats.last_sender_report_remote_timestamp_ms =
+        rtcp_sr_stats->last_remote_timestamp.ToMs();
+    stats.sender_reports_packets_sent = rtcp_sr_stats->packets_sent;
+    stats.sender_reports_bytes_sent = rtcp_sr_stats->bytes_sent;
+    stats.sender_reports_reports_count = rtcp_sr_stats->reports_count;
+  }
   return stats;
 }