RTCMediaStream[Track]Stats added.

webrtc/api/rtcstatscollector.cc

 /*
  *  Copyright 2016 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.
  */
 
 #include "webrtc/api/rtcstatscollector.h"
 
 #include <memory>
 #include <utility>
 #include <vector>
 
 #include "webrtc/api/peerconnection.h"
+#include "webrtc/api/peerconnectioninterface.h"
+#include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/webrtcsession.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/media/base/mediachannel.h"
 #include "webrtc/p2p/base/candidate.h"
 #include "webrtc/p2p/base/p2pconstants.h"
 #include "webrtc/p2p/base/port.h"
 
 namespace webrtc {
 
 namespace {
 
 std::string RTCCertificateIDFromFingerprint(const std::string& fingerprint) {
   return "RTCCertificate_" + fingerprint;
 }
 
 std::string RTCIceCandidatePairStatsIDFromConnectionInfo(
     const cricket::ConnectionInfo& info) {
   return "RTCIceCandidatePair_" + info.local_candidate.id() + "_" +
       info.remote_candidate.id();
 }
 
+std::string RTCMediaStreamTrackStatsIDFromMediaStreamTrackInterface(
+    const MediaStreamTrackInterface& track) {
+  return "RTCMediaStreamTrack_" + track.id();
+}
+
 std::string RTCTransportStatsIDFromTransportChannel(
     const std::string& transport_name, int channel_component) {
   return "RTCTransport_" + transport_name + "_" +
       rtc::ToString<>(channel_component);
 }
 
 std::string RTCTransportStatsIDFromBaseChannel(
     const ProxyTransportMap& proxy_to_transport,
     const cricket::BaseChannel& base_channel) {
   auto proxy_it = proxy_to_transport.find(base_channel.content_name());
@@ skipping 36 matching lines @@
     case DataChannelInterface::kClosing:
       return RTCDataChannelState::kClosing;
     case DataChannelInterface::kClosed:
       return RTCDataChannelState::kClosed;
     default:
       RTC_NOTREACHED();
       return nullptr;
   }
 }
 
+void SetMediaStreamTrackStatsFromMediaStreamTrackInterface(
+    const MediaStreamTrackInterface& track,
+    RTCMediaStreamTrackStats* track_stats) {
+  track_stats->track_identifier = track.id();
+  track_stats->ended = (track.state() == MediaStreamTrackInterface::kEnded);
+}
+
 void SetInboundRTPStreamStatsFromMediaReceiverInfo(
     const cricket::MediaReceiverInfo& media_receiver_info,
     RTCInboundRTPStreamStats* inbound_stats) {
   RTC_DCHECK(inbound_stats);
   inbound_stats->ssrc = rtc::ToString<>(media_receiver_info.ssrc());
   // TODO(hbos): Support the remote case. crbug.com/657855
   inbound_stats->is_remote = false;
   // TODO(hbos): Set |codec_id| when we have |RTCCodecStats|. Maybe relevant:
   // |media_receiver_info.codec_name|. crbug.com/657854, 657855, 659117
   inbound_stats->packets_received =
@@ skipping 101 matching lines @@
     candidate_stats->priority = static_cast<int32_t>(candidate.priority());
 
     stats = candidate_stats.get();
     report->AddStats(std::move(candidate_stats));
   }
   RTC_DCHECK_EQ(stats->type(), is_local ? RTCLocalIceCandidateStats::kType
                                         : RTCRemoteIceCandidateStats::kType);
   return stats->id();
 }
 
+void ProduceMediaStreamAndTrackStats(
+    int64_t timestamp_us,
+    rtc::scoped_refptr<StreamCollectionInterface> streams,
+    bool is_local,
+    RTCStatsReport* report) {
+  // TODO(hbos): This gets the current stream/track stats. The spec says that
+  // stream/track stats should appear in the stats as soon as they have been
+  // attached. This means we need to invalidate the collector's cache upon
+  // attaching or in some other way make sure that the new streams/tracks are
+  // accounted for. crbug.com/659137
+  // TODO(hbos): Also return detached tracks' stats (no longer present when
+  // iterating). The spec says the state should be a snapshot of the track at
+  // the moment of detachment. This means we have to either keep the track as-is
+  // after detachment or perform stats collection upon detachment.
+  // crbug.com/659137

[hbos, 2016/11/02 11:54:44]

Is it not the case that a stream can be detached as well? It has no "detached"
state and the comment about detached tracks stats continuing to appear are not
made for streams in the spec.

[hta-webrtc, 2016/11/03 14:34:21]

The spec has strongly deemphasized streams. In fact, we don't have a means of
detaching streams that come in using AddTrack - we're just assuming that they
drop off when the tracks are removed - and when AddTrack lands, the correct
iteration is to go through all tracks and take note of the streams in passing,
no tthe other way around.

But AddTrack isn't implemented yet.

I think the current code is fine for now.

[hbos, 2016/11/03 18:10:30]

Acknowledged.

+  if (!streams)
+    return;
+  for (size_t i = 0; i < streams->count(); ++i) {
+    MediaStreamInterface* stream = streams->at(i);
+
+    std::unique_ptr<RTCMediaStreamStats> stream_stats(
+        new RTCMediaStreamStats("RTCMediaStream_" + stream->label(),
+                                timestamp_us));
+    stream_stats->stream_identifier = stream->label();
+    stream_stats->track_ids = std::vector<std::string>();
+    // Audio Tracks
+    for (rtc::scoped_refptr<AudioTrackInterface> audio_track :
+         stream->GetAudioTracks()) {
+      std::unique_ptr<RTCMediaStreamTrackStats> audio_track_stats(
+          new RTCMediaStreamTrackStats(
+              RTCMediaStreamTrackStatsIDFromMediaStreamTrackInterface(
+                  *audio_track.get()),
+              timestamp_us));
+      stream_stats->track_ids->push_back(audio_track_stats->id());

[hta-webrtc, 2016/11/03 14:34:21]

Consider what happens here if a track is a member of two streams.

[hbos, 2016/11/03 18:10:30]

Done. Skipping if already exists.

+      SetMediaStreamTrackStatsFromMediaStreamTrackInterface(
+          *audio_track.get(),
+          audio_track_stats.get());
+      audio_track_stats->remote_source = !is_local;
+      audio_track_stats->detached = false;
+      int signal_level;
+      if (audio_track->GetSignalLevel(&signal_level)) {
+        // Convert signal level from [0,32767] int to [0,1] double.
+        RTC_DCHECK_GE(signal_level, 0);
+        RTC_DCHECK_LE(signal_level, 32767);
+        audio_track_stats->audio_level = signal_level / 32767.0;

[hta-webrtc, 2016/11/03 14:34:21]

Check this one with Niklas. Audio levels are awful stuff.
(and I find the description in the stats doc to be clear as mud, given that it
references a document that uses 0..127 on a logarithmic scale, while we want
0..1 on a linear one....)

[hbos, 2016/11/03 18:10:30]

+niklase: Is this the correct way to get the audio level?

+      }
+      if (audio_track->GetAudioProcessor()) {
+        AudioProcessorInterface::AudioProcessorStats audio_processor_stats;
+        audio_track->GetAudioProcessor()->GetStats(&audio_processor_stats);
+        audio_track_stats->echo_return_loss = static_cast<double>(
+            audio_processor_stats.echo_return_loss);
+        audio_track_stats->echo_return_loss_enhancement = static_cast<double>(
+            audio_processor_stats.echo_return_loss_enhancement);
+      }
+      report->AddStats(std::move(audio_track_stats));
+    }
+    // Video Tracks
+    for (rtc::scoped_refptr<VideoTrackInterface> video_track :
+         stream->GetVideoTracks()) {
+      std::unique_ptr<RTCMediaStreamTrackStats> video_track_stats(
+          new RTCMediaStreamTrackStats(
+              RTCMediaStreamTrackStatsIDFromMediaStreamTrackInterface(
+                  *video_track.get()),
+              timestamp_us));
+      stream_stats->track_ids->push_back(video_track_stats->id());
+      SetMediaStreamTrackStatsFromMediaStreamTrackInterface(
+          *video_track.get(),
+          video_track_stats.get());
+      video_track_stats->remote_source = !is_local;
+      video_track_stats->detached = false;
+      if (video_track->GetSource()) {
+        VideoTrackSourceInterface::Stats video_track_source_stats;
+        if (video_track->GetSource()->GetStats(&video_track_source_stats)) {
+          video_track_stats->frame_width = static_cast<uint32_t>(
+              video_track_source_stats.input_width);
+          video_track_stats->frame_height = static_cast<uint32_t>(
+              video_track_source_stats.input_height);

[hbos, 2016/11/02 11:44:55]

I wonder if this is the right value. What if the source is a HD video but the
output is downscaled, which value should the track have?

[hbos, 2016/11/02 11:48:41]

And there are similar stats to the spec in Video[Sender/Receiver]Info but that
is currently not accessible from the stream/track?

[hta-webrtc, 2016/11/03 14:34:21]

Yes. More work needed both for getSettings and for these stats.

[hta-webrtc, 2016/11/03 14:34:21]

It should be the downscaled value. I have a TODO to fix width and height to
reflect the downscaled value for getSettings(), but haven't gotten around to it
yet.

[hbos, 2016/11/03 18:10:30]

What are you referring to when you say "getSettings"?

[hta-webrtc, 2016/11/04 16:30:01]

https://w3c.github.io/mediacapture-main/getusermedia.html#dom-mediastreamtrac...

+        }
+      }
+      report->AddStats(std::move(video_track_stats));
+    }
+    report->AddStats(std::move(stream_stats));
+  }
+}
+
 }  // namespace
 
 rtc::scoped_refptr<RTCStatsCollector> RTCStatsCollector::Create(
     PeerConnection* pc, int64_t cache_lifetime_us) {
   return rtc::scoped_refptr<RTCStatsCollector>(
       new rtc::RefCountedObject<RTCStatsCollector>(pc, cache_lifetime_us));
 }
 
 RTCStatsCollector::RTCStatsCollector(PeerConnection* pc,
                                      int64_t cache_lifetime_us)
@@ skipping 67 matching lines @@
     ProduceCertificateStats_s(
         timestamp_us, transport_cert_stats, report.get());
     ProduceIceCandidateAndPairStats_s(
         timestamp_us, session_stats, report.get());
     ProduceRTPStreamStats_s(
         timestamp_us, session_stats, report.get());
     ProduceTransportStats_s(
         timestamp_us, session_stats, transport_cert_stats, report.get());
   }
   ProduceDataChannelStats_s(timestamp_us, report.get());
+  ProduceMediaStreamAndTrackStats_s(timestamp_us, report.get());
   ProducePeerConnectionStats_s(timestamp_us, report.get());
 
   AddPartialResults(report);
 }
 
 void RTCStatsCollector::ProducePartialResultsOnWorkerThread(
     int64_t timestamp_us) {
   RTC_DCHECK(worker_thread_->IsCurrent());
   rtc::scoped_refptr<RTCStatsReport> report = RTCStatsReport::Create(
       timestamp_us);
@@ skipping 146 matching lines @@
             static_cast<uint64_t>(info.recv_ping_responses);
         candidate_pair_stats->responses_sent =
             static_cast<uint64_t>(info.sent_ping_responses);
 
         report->AddStats(std::move(candidate_pair_stats));
       }
     }
   }
 }
 
+void RTCStatsCollector::ProduceMediaStreamAndTrackStats_s(
+    int64_t timestamp_us, RTCStatsReport* report) const {
+  RTC_DCHECK(signaling_thread_->IsCurrent());
+  ProduceMediaStreamAndTrackStats(
+      timestamp_us, pc_->local_streams(), true, report);
+  ProduceMediaStreamAndTrackStats(
+      timestamp_us, pc_->remote_streams(), false, report);
+}
+
 void RTCStatsCollector::ProducePeerConnectionStats_s(
     int64_t timestamp_us, RTCStatsReport* report) const {
   RTC_DCHECK(signaling_thread_->IsCurrent());
   // TODO(hbos): If data channels are removed from the peer connection this will
   // yield incorrect counts. Address before closing crbug.com/636818. See
   // https://w3c.github.io/webrtc-stats/webrtc-stats.html#pcstats-dict*.
   uint32_t data_channels_opened = 0;
   const std::vector<rtc::scoped_refptr<DataChannel>>& data_channels =
       pc_->sctp_data_channels();
   for (const rtc::scoped_refptr<DataChannel>& data_channel : data_channels) {
@@ skipping 199 matching lines @@
     const std::string& type) {
   return CandidateTypeToRTCIceCandidateType(type);
 }
 
 const char* DataStateToRTCDataChannelStateForTesting(
     DataChannelInterface::DataState state) {
   return DataStateToRTCDataChannelState(state);
 }
 
 }  // namespace webrtc