Get VideoCapturer stats via VideoTrackSourceInterface in StatsCollector,

webrtc/api/statscollector.cc

Index: webrtc/api/statscollector.cc
diff --git a/webrtc/api/statscollector.cc b/webrtc/api/statscollector.cc
index d77953b3bed1d49bb75a3be6a35c5d10f91c382b..cd9e1899ebfc04ff3234dae9eff25c37ca8413a6 100644
--- a/webrtc/api/statscollector.cc
+++ b/webrtc/api/statscollector.cc
@@ -236,11 +236,9 @@ void ExtractStats(const cricket::VideoSenderInfo& info, StatsReport* report) {
     { StatsReport::kStatsValueNameEncodeUsagePercent,
       info.encode_usage_percent },
     { StatsReport::kStatsValueNameFirsReceived, info.firs_rcvd },
-    { StatsReport::kStatsValueNameFrameHeightInput, info.input_frame_height },
     { StatsReport::kStatsValueNameFrameHeightSent, info.send_frame_height },
     { StatsReport::kStatsValueNameFrameRateInput, info.framerate_input },
     { StatsReport::kStatsValueNameFrameRateSent, info.framerate_sent },
-    { StatsReport::kStatsValueNameFrameWidthInput, info.input_frame_width },
     { StatsReport::kStatsValueNameFrameWidthSent, info.send_frame_width },
     { StatsReport::kStatsValueNameNacksReceived, info.nacks_rcvd },
     { StatsReport::kStatsValueNamePacketsLost, info.packets_lost },
@@ -474,6 +472,7 @@ StatsCollector::UpdateStats(PeerConnectionInterface::StatsOutputLevel level) {
     ExtractSessionInfo();
     ExtractVoiceInfo();
     ExtractVideoInfo(level);
+    ExtractSenderInfo();
     ExtractDataInfo();
     UpdateTrackReports();
   }
@@ -828,6 +827,32 @@ void StatsCollector::ExtractVideoInfo(
   }
 }
 
+void StatsCollector::ExtractSenderInfo() {
+  RTC_DCHECK(pc_->session()->signaling_thread()->IsCurrent());
+
+  for (const auto &sender : pc_->GetSenders()) {

[pthatcher1, 2016/03/30 16:48:43]

const auto& sender

[nisse-webrtc, 2016/03/31 06:31:22]

Done.

+    const rtc::scoped_refptr<MediaStreamTrackInterface> track(sender->track());
+    if (!track || track->kind() != MediaStreamTrackInterface::kVideoKind)

[Taylor Brandstetter, 2016/03/30 17:42:32]

nit: {}s

[nisse-webrtc, 2016/03/31 06:31:22]

Done.

+      continue;
+
+    VideoTrackSourceInterface *source =

[pthatcher1, 2016/03/30 16:48:43]

VideoTrackSourceInterface* source

[nisse-webrtc, 2016/03/31 06:31:22]

Done.

+      static_cast<VideoTrackInterface*>(track.get())->GetSource();

[pthatcher1, 2016/03/30 16:48:43]

I agree with the decision that you and Tommi made to keep this logic here. 
However, if we're going to be relying on the fact that any
MediaStreamTrackInterface with kind == MediaStreamTrackInterface::kVideoKind is
really a VideoTrackInterface, we should document that clearly in two places:

1.  In mediastreaminterface.h, where MediaStreamTrackInterface is defined.

2.  Here, saying "this is safe because ...; see mediastreaminterface.h".

[nisse-webrtc, 2016/03/31 06:31:22]

Done. We're not using dynamic_cast<>? This must be the type of situation it is
intended for.

+
+    VideoTrackSourceInterface::Stats stats;
+    if (source->GetStats(&stats)) {

[pthatcher1, 2016/03/30 16:48:43]

I'd prefer:

 if (!source->GetStats(&stats)) {
  continue:
}

...

[nisse-webrtc, 2016/03/31 06:31:22]

Done.

+      const StatsReport::Id stats_id =
+        StatsReport::NewIdWithDirection(StatsReport::kStatsReportTypeSsrc,
+                                        rtc::ToString<uint32_t>(sender->ssrc()),

[Taylor Brandstetter, 2016/03/30 17:42:32]

It's possible that a sender doesn't have an SSRC (ssrc() == 0), if no local
description has been set yet. In that case it doesn't make sense to report these
stats, since they hinge off an SSRC. So you should add "|| sender->ssrc() == 0"
to the above "if" statement.

[nisse-webrtc, 2016/03/31 06:31:22]

I added it as a separate "if"; no need to fiddle with the track refcount if
we're not going to use it.

My understanding is that ssrc == 0 is technically valid in the webrtc wire
protocols, so this logic is something we'd like to change eventually?

[Taylor Brandstetter, 2016/03/31 18:25:03]

We should change it to an rtc::Optional in that case. It would make things more
clear anyway. I'll assign a bug to myself to remind myself to do this.

[nisse-webrtc, 2016/04/01 06:28:28]

If possible, it would be even clearer if ssrc is always set, or always set if a
track is set. Is it reasonable to attach a track before the ssrc is known?

[Taylor Brandstetter, 2016/04/01 18:28:40]

Yes, it is reasonable. Through the API you can use "addTransceiver" to create a
sender, and then add a track to it, before any SDP negotiation (which decides
SSRCs) has been done.

+                                        StatsReport::kSend);
+      StatsReport* report = reports_.FindOrAddNew(stats_id);
+      report->AddInt(StatsReport::kStatsValueNameFrameWidthInput,
+                     stats.input_width);
+      report->AddInt(StatsReport::kStatsValueNameFrameHeightInput,
+                     stats.input_height);
+    }
+  }
+}
+
 void StatsCollector::ExtractDataInfo() {
   RTC_DCHECK(pc_->session()->signaling_thread()->IsCurrent());