Reland "Add hugeFramesSent GetStats metric"

This is a reland of f9f71b91ae073fdd2b89ff9df1204835aa3137eb
after the change in chromium tests.

Chromium change done here:
https://chromium-review.googlesource.com/c/chromium/src/+/950776

Original reviewed on: https://webrtc-review.googlesource.com/c/src/+/54420

No changes to the original patchset were done.

TBR=hta@webrtc.org,hbos@webrtc.org,sprang@webrtc.org,solenberg@webrtc.org

Bug: webrtc:8901
Change-Id: Ic88c3cb963dceea0426eb90519743e3c1a4533c1
Reviewed-on: https://webrtc-review.googlesource.com/60140
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22310}

api/stats/rtcstats_objects.h

@@ -279,6 +279,7 @@ class RTCMediaStreamTrackStats final : public RTCStats {
   // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/659137
   RTCStatsMember<double> frames_per_second;
   RTCStatsMember<uint32_t> frames_sent;
+  RTCStatsMember<uint32_t> huge_frames_sent;
   RTCStatsMember<uint32_t> frames_received;
   RTCStatsMember<uint32_t> frames_decoded;
   RTCStatsMember<uint32_t> frames_dropped;

api/statstypes.cc

@@ -558,6 +558,8 @@ const char* StatsReport::Value::display_name() const {
       return "googFrameWidthSent";
     case kStatsValueNameHasEnteredLowResolution:
       return "googHasEnteredLowResolution";
+    case kStatsValueNameHugeFramesSent:
+      return "hugeFramesSent";
     case kStatsValueNameInitiator:
       return "googInitiator";
     case kStatsValueNameInterframeDelayMaxMs:

api/statstypes.h

@@ -189,6 +189,7 @@ class StatsReport {
     kStatsValueNameFrameWidthReceived,
     kStatsValueNameFrameWidthSent,
     kStatsValueNameHasEnteredLowResolution,
+    kStatsValueNameHugeFramesSent,
     kStatsValueNameInitiator,
     kStatsValueNameInterframeDelayMaxMs,  // Max over last 10 seconds.
     kStatsValueNameIssuerId,

call/video_send_stream.h

@@ -91,6 +91,7 @@ class VideoSendStream {
     std::map<uint32_t, StreamStats> substreams;
     webrtc::VideoContentType content_type =
         webrtc::VideoContentType::UNSPECIFIED;
+    uint32_t huge_frames_sent = 0;
   };
 
   struct Config {

media/base/mediachannel.h

@@ -466,6 +466,8 @@ struct VideoSenderInfo : public MediaSenderInfo {
   bool has_entered_low_resolution = false;
   rtc::Optional<uint64_t> qp_sum;
   webrtc::VideoContentType content_type = webrtc::VideoContentType::UNSPECIFIED;
+  // https://w3c.github.io/webrtc-stats/#dom-rtcvideosenderstats-hugeframessent
+  uint32_t huge_frames_sent = 0;
 };
 
 struct VideoReceiverInfo : public MediaReceiverInfo {

media/engine/webrtcvideoengine.cc

@@ -2058,6 +2058,7 @@ VideoSenderInfo WebRtcVideoChannel::WebRtcVideoSendStream::GetVideoSenderInfo(
   info.preferred_bitrate = stats.preferred_media_bitrate_bps;
 
   info.content_type = stats.content_type;
+  info.huge_frames_sent = stats.huge_frames_sent;
 
   info.send_frame_width = 0;
   info.send_frame_height = 0;

pc/rtcstats_integrationtest.cc

@@ -515,6 +515,7 @@ class RTCStatsReportVerifier {
       verifier.TestMemberIsUndefined(media_stream_track.frames_per_second);
       if (*media_stream_track.remote_source) {
         verifier.TestMemberIsUndefined(media_stream_track.frames_sent);
+        verifier.TestMemberIsUndefined(media_stream_track.huge_frames_sent);
         verifier.TestMemberIsNonNegative<uint32_t>(
             media_stream_track.frames_received);
         verifier.TestMemberIsNonNegative<uint32_t>(
@@ -524,6 +525,8 @@ class RTCStatsReportVerifier {
       } else {
         verifier.TestMemberIsNonNegative<uint32_t>(
             media_stream_track.frames_sent);
+        verifier.TestMemberIsNonNegative<uint32_t>(
+            media_stream_track.huge_frames_sent);
         verifier.TestMemberIsUndefined(media_stream_track.frames_received);
         verifier.TestMemberIsUndefined(media_stream_track.frames_decoded);
         verifier.TestMemberIsUndefined(media_stream_track.frames_dropped);
@@ -546,6 +549,7 @@ class RTCStatsReportVerifier {
       verifier.TestMemberIsUndefined(media_stream_track.frame_height);
       verifier.TestMemberIsUndefined(media_stream_track.frames_per_second);
       verifier.TestMemberIsUndefined(media_stream_track.frames_sent);
+      verifier.TestMemberIsUndefined(media_stream_track.huge_frames_sent);
       verifier.TestMemberIsUndefined(media_stream_track.frames_received);
       verifier.TestMemberIsUndefined(media_stream_track.frames_decoded);
       verifier.TestMemberIsUndefined(media_stream_track.frames_dropped);

pc/rtcstatscollector.cc

@@ -455,6 +455,7 @@ ProduceMediaStreamTrackStatsFromVideoSenderInfo(
   // TODO(hbos): Will reduce this by frames dropped due to congestion control
   // when available. https://crbug.com/659137
   video_track_stats->frames_sent = video_sender_info.frames_encoded;
+  video_track_stats->huge_frames_sent = video_sender_info.huge_frames_sent;
   return video_track_stats;
 }
 

pc/rtcstatscollector_unittest.cc

@@ -1333,6 +1333,7 @@ TEST_F(RTCStatsCollectorTest,
   video_sender_info_ssrc1.send_frame_width = 1234;
   video_sender_info_ssrc1.send_frame_height = 4321;
   video_sender_info_ssrc1.frames_encoded = 11;
+  video_sender_info_ssrc1.huge_frames_sent = 1;
 
   stats_->CreateMockRtpSendersReceiversAndChannels(
       {}, {},
@@ -1367,6 +1368,7 @@ TEST_F(RTCStatsCollectorTest,
   expected_local_video_track_ssrc1.frame_width = 1234;
   expected_local_video_track_ssrc1.frame_height = 4321;
   expected_local_video_track_ssrc1.frames_sent = 11;
+  expected_local_video_track_ssrc1.huge_frames_sent = 1;
   ASSERT_TRUE(report->Get(expected_local_video_track_ssrc1.id()));
   EXPECT_EQ(expected_local_video_track_ssrc1,
             report->Get(expected_local_video_track_ssrc1.id())

pc/statscollector.cc

@@ -313,20 +313,21 @@ void ExtractStats(const cricket::VideoSenderInfo& info, StatsReport* report) {
     report->AddInt(StatsReport::kStatsValueNameQpSum, *info.qp_sum);
 
   const IntForAdd ints[] = {
-    { StatsReport::kStatsValueNameAdaptationChanges, info.adapt_changes },
+      {StatsReport::kStatsValueNameAdaptationChanges, info.adapt_changes},
-    { StatsReport::kStatsValueNameAvgEncodeMs, info.avg_encode_ms },
+      {StatsReport::kStatsValueNameAvgEncodeMs, info.avg_encode_ms},
-    { StatsReport::kStatsValueNameEncodeUsagePercent,
+      {StatsReport::kStatsValueNameEncodeUsagePercent,
-      info.encode_usage_percent },
+       info.encode_usage_percent},
-    { StatsReport::kStatsValueNameFirsReceived, info.firs_rcvd },
+      {StatsReport::kStatsValueNameFirsReceived, info.firs_rcvd},
-    { StatsReport::kStatsValueNameFrameHeightSent, info.send_frame_height },
+      {StatsReport::kStatsValueNameFrameHeightSent, info.send_frame_height},
-    { StatsReport::kStatsValueNameFrameRateInput, info.framerate_input },
+      {StatsReport::kStatsValueNameFrameRateInput, info.framerate_input},
-    { StatsReport::kStatsValueNameFrameRateSent, info.framerate_sent },
+      {StatsReport::kStatsValueNameFrameRateSent, info.framerate_sent},
-    { StatsReport::kStatsValueNameFrameWidthSent, info.send_frame_width },
+      {StatsReport::kStatsValueNameFrameWidthSent, info.send_frame_width},
-    { StatsReport::kStatsValueNameNacksReceived, info.nacks_rcvd },
+      {StatsReport::kStatsValueNameNacksReceived, info.nacks_rcvd},
-    { StatsReport::kStatsValueNamePacketsLost, info.packets_lost },
+      {StatsReport::kStatsValueNamePacketsLost, info.packets_lost},
-    { StatsReport::kStatsValueNamePacketsSent, info.packets_sent },
+      {StatsReport::kStatsValueNamePacketsSent, info.packets_sent},
-    { StatsReport::kStatsValueNamePlisReceived, info.plis_rcvd },
+      {StatsReport::kStatsValueNamePlisReceived, info.plis_rcvd},
-    { StatsReport::kStatsValueNameFramesEncoded, info.frames_encoded },
+      {StatsReport::kStatsValueNameFramesEncoded, info.frames_encoded},
+      {StatsReport::kStatsValueNameHugeFramesSent, info.huge_frames_sent},
   };
 
   for (const auto& i : ints)

stats/rtcstats_objects.cc

@@ -383,6 +383,7 @@ WEBRTC_RTCSTATS_IMPL(RTCMediaStreamTrackStats, RTCStats, "track",
                      &frame_height,
                      &frames_per_second,
                      &frames_sent,
+                     &huge_frames_sent,
                      &frames_received,
                      &frames_decoded,
                      &frames_dropped,
@@ -418,6 +419,7 @@ RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(std::string&& id,
       frame_height("frameHeight"),
       frames_per_second("framesPerSecond"),
       frames_sent("framesSent"),
+      huge_frames_sent("hugeFramesSent"),
       frames_received("framesReceived"),
       frames_decoded("framesDecoded"),
       frames_dropped("framesDropped"),
@@ -449,6 +451,7 @@ RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(
       frame_height(other.frame_height),
       frames_per_second(other.frames_per_second),
       frames_sent(other.frames_sent),
+      huge_frames_sent(other.huge_frames_sent),
       frames_received(other.frames_received),
       frames_decoded(other.frames_decoded),
       frames_dropped(other.frames_dropped),

video/send_statistics_proxy.cc

@@ -914,6 +914,17 @@ void SendStatisticsProxy::OnSendEncodedImage(
     }
   }
 
+  // If any of the simulcast streams have a huge frame, it should be counted
+  // as a single difficult input frame.
+  // https://w3c.github.io/webrtc-stats/#dom-rtcvideosenderstats-hugeframessent
+  if (encoded_image.timing_.flags & TimingFrameFlags::kTriggeredBySize) {
+    if (!last_outlier_timestamp_ ||
+        *last_outlier_timestamp_ < encoded_image.capture_time_ms_) {
+      last_outlier_timestamp_.emplace(encoded_image.capture_time_ms_);
+      ++stats_.huge_frames_sent;
+    }
+  }
+
   media_byte_rate_tracker_.AddSamples(encoded_image._length);
 
   // Initialize to current since |is_limited_in_resolution| is only updated

video/send_statistics_proxy.h

@@ -249,6 +249,8 @@ class SendStatisticsProxy : public CpuOveruseMetricsObserver,
   rtc::RateTracker media_byte_rate_tracker_ RTC_GUARDED_BY(crit_);
   rtc::RateTracker encoded_frame_rate_tracker_ RTC_GUARDED_BY(crit_);
 
+  rtc::Optional<int64_t> last_outlier_timestamp_ RTC_GUARDED_BY(crit_);
+
   // Contains stats used for UMA histograms. These stats will be reset if
   // content type changes between real-time video and screenshare, since these
   // will be reported separately.