Get VideoCapturer stats via VideoTrackSourceInterface in StatsCollector,

webrtc/api/statscollector.cc

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 218 matching lines @@
                      (info.adapt_reason & 0x1) > 0);
   report->AddBoolean(StatsReport::kStatsValueNameViewLimitedResolution,
                      (info.adapt_reason & 0x4) > 0);
 
   const IntForAdd ints[] = {
     { StatsReport::kStatsValueNameAdaptationChanges, info.adapt_changes },
     { StatsReport::kStatsValueNameAvgEncodeMs, info.avg_encode_ms },
     { 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 },
     { StatsReport::kStatsValueNamePacketsSent, info.packets_sent },
     { StatsReport::kStatsValueNamePlisReceived, info.plis_rcvd },
   };
 
   for (const auto& i : ints)
     report->AddInt(i.name, i.value);
   report->AddString(StatsReport::kStatsValueNameMediaType, "video");
@@ skipping 213 matching lines @@
   if (pc_->session()) {
     // TODO(tommi): All of these hop over to the worker thread to fetch
     // information.  We could use an AsyncInvoker to run all of these and post
     // the information back to the signaling thread where we can create and
     // update stats reports.  That would also clean up the threading story a bit
     // since we'd be creating/updating the stats report objects consistently on
     // the same thread (this class has no locks right now).
     ExtractSessionInfo();
     ExtractVoiceInfo();
     ExtractVideoInfo(level);
+    ExtractSenderInfo();
     ExtractDataInfo();
     UpdateTrackReports();
   }
 }
 
 StatsReport* StatsCollector::PrepareReport(
     bool local,
     uint32_t ssrc,
     const StatsReport::Id& transport_id,
     StatsReport::Direction direction) {
@@ skipping 334 matching lines @@
   if (video_info.bw_estimations.size() != 1) {
     LOG(LS_ERROR) << "BWEs count: " << video_info.bw_estimations.size();
   } else {
     StatsReport::Id report_id(StatsReport::NewBandwidthEstimationId());
     StatsReport* report = reports_.FindOrAddNew(report_id);
     ExtractStats(
         video_info.bw_estimations[0], stats_gathering_started_, level, report);
   }
 }
 
+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());
 
   rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
 
   for (const auto& dc : pc_->sctp_data_channels()) {
     StatsReport::Id id(StatsReport::NewTypedIntId(
         StatsReport::kStatsReportTypeDataChannel, dc->id()));
     StatsReport* report = reports_.ReplaceOrAddNew(id);
     report->set_timestamp(stats_gathering_started_);
@@ skipping 97 matching lines @@
     StatsReport* report = entry.second;
     report->set_timestamp(stats_gathering_started_);
   }
 }
 
 void StatsCollector::ClearUpdateStatsCacheForTest() {
   stats_gathering_started_ = 0;
 }
 
 }  // namespace webrtc