[Stats] Add googTimingFrameInfo to the modern API.

This is exposing something that is already exposed in the legacy
getStats() API and is only available if the "video-timing" header
extension is used. Adding this metric here should unblock legacy
getStats() API deprecation. The follow-up to unship or standardize this
metric is tracked by https://crbug.com/webrtc/14586.

Bug: webrtc:14587
Change-Id: Ic3d45b0558d7caf4be2856a4cd95b88db312f85e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279860
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38444}

api/stats/rtcstats_objects.h

@@ -502,6 +502,13 @@ class RTC_EXPORT RTCInboundRTPStreamStats final
   RTCStatsMember<uint32_t> pli_count;
   RTCStatsMember<uint32_t> nack_count;
   RTCStatsMember<uint64_t> qp_sum;
+  // This is a remnant of the legacy getStats() API. When the "video-timing"
+  // header extension is used,
+  // https://webrtc.github.io/webrtc-org/experiments/rtp-hdrext/video-timing/,
+  // `googTimingFrameInfo` is exposed with the value of
+  // TimingFrameInfo::ToString().
+  // TODO(https://crbug.com/webrtc/14586): Unship or standardize this metric.
+  RTCStatsMember<std::string> goog_timing_frame_info;
   // Non-standard audio metrics.
   RTCNonStandardStatsMember<uint64_t> jitter_buffer_flushes;
   RTCNonStandardStatsMember<uint64_t> delayed_packet_outage_samples;

pc/rtc_stats_collector.cc

@@ -607,8 +607,13 @@ void SetInboundRTPStreamStatsFromVideoReceiverInfo(
   if (video_receiver_info.framerate_decoded > 0) {
     inbound_video->frames_per_second = video_receiver_info.framerate_decoded;
   }
-  if (video_receiver_info.qp_sum)
+  if (video_receiver_info.qp_sum) {
     inbound_video->qp_sum = *video_receiver_info.qp_sum;
+  }
+  if (video_receiver_info.timing_frame_info.has_value()) {
+    inbound_video->goog_timing_frame_info =
+        video_receiver_info.timing_frame_info->ToString();
+  }
   inbound_video->total_decode_time =
       video_receiver_info.total_decode_time.seconds<double>();
   inbound_video->total_processing_delay =

pc/rtc_stats_collector_unittest.cc

@@ -38,6 +38,7 @@
 #include "api/video/video_frame.h"
 #include "api/video/video_sink_interface.h"
 #include "api/video/video_source_interface.h"
+#include "api/video/video_timing.h"
 #include "common_video/include/quality_limitation_reason.h"
 #include "media/base/media_channel.h"
 #include "modules/audio_processing/include/audio_processing_statistics.h"
@@ -2714,6 +2715,42 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
   EXPECT_TRUE(report->Get(*expected_video.codec_id));
 }
 
+TEST_F(RTCStatsCollectorTest, CollectGoogTimingFrameInfo) {
+  cricket::VideoMediaInfo video_media_info;
+
+  video_media_info.receivers.push_back(cricket::VideoReceiverInfo());
+  video_media_info.receivers[0].local_stats.push_back(
+      cricket::SsrcReceiverInfo());
+  video_media_info.receivers[0].local_stats[0].ssrc = 1;
+  TimingFrameInfo timing_frame_info;
+  timing_frame_info.rtp_timestamp = 1;
+  timing_frame_info.capture_time_ms = 2;
+  timing_frame_info.encode_start_ms = 3;
+  timing_frame_info.encode_finish_ms = 4;
+  timing_frame_info.packetization_finish_ms = 5;
+  timing_frame_info.pacer_exit_ms = 6;
+  timing_frame_info.network_timestamp_ms = 7;
+  timing_frame_info.network2_timestamp_ms = 8;
+  timing_frame_info.receive_start_ms = 9;
+  timing_frame_info.receive_finish_ms = 10;
+  timing_frame_info.decode_start_ms = 11;
+  timing_frame_info.decode_finish_ms = 12;
+  timing_frame_info.render_time_ms = 13;
+  timing_frame_info.flags = 14;
+  video_media_info.receivers[0].timing_frame_info = timing_frame_info;
+
+  pc_->AddVideoChannel("Mid0", "Transport0", video_media_info);
+  stats_->SetupRemoteTrackAndReceiver(
+      cricket::MEDIA_TYPE_VIDEO, "RemoteVideoTrackID", "RemoteStreamId", 1);
+
+  rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
+  auto inbound_rtps = report->GetStatsOfType<RTCInboundRTPStreamStats>();
+  ASSERT_EQ(inbound_rtps.size(), 1u);
+  ASSERT_TRUE(inbound_rtps[0]->goog_timing_frame_info.is_defined());
+  EXPECT_EQ(*inbound_rtps[0]->goog_timing_frame_info,
+            "1,2,3,4,5,6,7,8,9,10,11,12,13,1,0");
+}
+
 TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) {
   cricket::VoiceMediaInfo voice_media_info;
 

pc/rtc_stats_integrationtest.cc

@@ -937,6 +937,7 @@ class RTCStatsReportVerifier {
           inbound_stream.total_interruption_duration);
       verifier.TestMemberIsNonNegative<double>(
           inbound_stream.min_playout_delay);
+      verifier.TestMemberIsDefined(inbound_stream.goog_timing_frame_info);
     } else {
       verifier.TestMemberIsUndefined(inbound_stream.frames_decoded);
       verifier.TestMemberIsUndefined(inbound_stream.key_frames_decoded);
@@ -965,6 +966,7 @@ class RTCStatsReportVerifier {
       verifier.TestMemberIsNonNegative<double>(
           inbound_stream.total_interruption_duration);
       verifier.TestMemberIsUndefined(inbound_stream.min_playout_delay);
+      verifier.TestMemberIsUndefined(inbound_stream.goog_timing_frame_info);
     }
     return verifier.ExpectAllMembersSuccessfullyTested();
   }

stats/rtcstats_objects.cc

@@ -585,6 +585,7 @@ WEBRTC_RTCSTATS_IMPL(
     &pli_count,
     &nack_count,
     &qp_sum,
+    &goog_timing_frame_info,
     &jitter_buffer_flushes,
     &delayed_packet_outage_samples,
     &relative_packet_arrival_delay,
@@ -648,6 +649,7 @@ RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(std::string&& id,
       pli_count("pliCount"),
       nack_count("nackCount"),
       qp_sum("qpSum"),
+      goog_timing_frame_info("googTimingFrameInfo"),
       jitter_buffer_flushes(
           "jitterBufferFlushes",
           {NonStandardGroupId::kRtcAudioJitterBufferMaxPackets}),