RTCOutboundRTPStreamStats 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/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 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());
+  if (proxy_it == proxy_to_transport.cend())
+    return "";
+  return RTCTransportStatsIDFromTransportChannel(
+      proxy_it->second, cricket::ICE_CANDIDATE_COMPONENT_RTP);
+}
+
+std::string RTCOutboundRTPStreamStatsIDFromSSRC(bool audio, uint32_t ssrc) {
+  return audio ? "RTCOutboundRTPAudioStream_" + rtc::ToString<>(ssrc)
+               : "RTCOutboundRTPVideoStream_" + rtc::ToString<>(ssrc);
+}
+
 const char* CandidateTypeToRTCIceCandidateType(const std::string& type) {
   if (type == cricket::LOCAL_PORT_TYPE)
     return RTCIceCandidateType::kHost;
   if (type == cricket::STUN_PORT_TYPE)
     return RTCIceCandidateType::kSrflx;
   if (type == cricket::PRFLX_PORT_TYPE)
     return RTCIceCandidateType::kPrflx;
   if (type == cricket::RELAY_PORT_TYPE)
     return RTCIceCandidateType::kRelay;
   RTC_NOTREACHED();
@@ skipping 10 matching lines @@
     case DataChannelInterface::kClosing:
       return RTCDataChannelState::kClosing;
     case DataChannelInterface::kClosed:
       return RTCDataChannelState::kClosed;
     default:
       RTC_NOTREACHED();
       return nullptr;
   }
 }
 
+void SetOutboundRTPStreamStatsFromMediaSenderInfo(
+    const cricket::MediaSenderInfo& media_sender_info,
+    RTCOutboundRTPStreamStats* outbound_stats) {
+  RTC_DCHECK(outbound_stats);
+  outbound_stats->ssrc = rtc::ToString<>(media_sender_info.ssrc());
+  // TODO(hbos): Support the remote case. crbug.com/657856
+  outbound_stats->is_remote = false;
+  // TODO(hbos): Set |codec_id| when we have |RTCCodecStats|. Maybe relevant:
+  // |media_sender_info.codec_name|. crbug.com/657854, 657856, 659117
+  outbound_stats->packets_sent =
+      static_cast<uint32_t>(media_sender_info.packets_sent);
+  outbound_stats->bytes_sent =
+      static_cast<uint64_t>(media_sender_info.bytes_sent);
+  outbound_stats->round_trip_time =
+      static_cast<double>(media_sender_info.rtt_ms) / rtc::kNumMillisecsPerSec;
+}
+
+void SetOutboundRTPStreamStatsFromVoiceSenderInfo(
+    const cricket::VoiceSenderInfo& voice_sender_info,
+    RTCOutboundRTPStreamStats* outbound_stats) {
+  SetOutboundRTPStreamStatsFromMediaSenderInfo(
+      voice_sender_info, outbound_stats);
+  outbound_stats->media_type = "audio";
+  // |fir_count|, |pli_count|, |nack_count| and |sli_count| are only valid for
+  // video and are purposefully left undefined for audio.

[hbos, 2016/10/26 10:26:11]

Is this true? Even for NACK?

FIR and SLI are documented as only valid for video.
PLI is said to be "packet loss indication" and is not documented as video-only,
but the reference link says it's "picture loss indication" and is about video
data.
NACK on the other hand has to do with RTP packets lost, is that not
video-specific?

[Taylor Brandstetter, 2016/10/27 21:44:36]

NACK seems relevant to audio, PLI doesn't. I'll make a stats PR to fix that.

[hbos, 2016/10/28 09:42:37]

Acknowledged.

+}
+
+void SetOutboundRTPStreamStatsFromVideoSenderInfo(
+    const cricket::VideoSenderInfo& video_sender_info,
+    RTCOutboundRTPStreamStats* outbound_stats) {
+  SetOutboundRTPStreamStatsFromMediaSenderInfo(
+      video_sender_info, outbound_stats);
+  outbound_stats->media_type = "video";
+  outbound_stats->fir_count =
+      static_cast<uint32_t>(video_sender_info.firs_rcvd);
+  outbound_stats->pli_count =
+      static_cast<uint32_t>(video_sender_info.plis_rcvd);
+  outbound_stats->nack_count =
+      static_cast<uint32_t>(video_sender_info.nacks_rcvd);
+  outbound_stats->target_bitrate =
+      static_cast<double>(video_sender_info.preferred_bitrate);

[hbos, 2016/10/26 10:26:11]

Is this the correct value? There is also bandwidth estimation info[1], perhaps
this should be target_enc_bitrate?

[1]
https://cs.chromium.org/chromium/src/third_party/webrtc/media/base/mediachann...

[Taylor Brandstetter, 2016/10/27 21:44:36]

I came to the same conclusion; it should be target_enc_bitrate.

[hbos, 2016/10/28 09:42:37]

Done. Note that this means that all video-RTCOutboundRTPStreamStats gets the
same target_bitrate value.

[Taylor Brandstetter, 2016/10/28 17:31:34]

Wait, that doesn't seem right. I didn't realize target_enc_bitrate was combined
for all streams; what we really want are the individual
"target_media_bitrate_bps" values that go into this:
https://cs.chromium.org/chromium/src/third_party/webrtc/video_send_stream.h?d...

If there's no way to get that currently, guess we need another TODO. Sorry for
my confusion.

[hbos, 2016/10/31 09:21:01]

That will be easy to surface but I'll do it in a separate CL, making it a TODO
here.

+}
+
 }  // 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 49 matching lines @@
 }
 
 void RTCStatsCollector::ClearCachedStatsReport() {
   RTC_DCHECK(signaling_thread_->IsCurrent());
   cached_report_ = nullptr;
 }
 
 void RTCStatsCollector::ProducePartialResultsOnSignalingThread(
     int64_t timestamp_us) {
   RTC_DCHECK(signaling_thread_->IsCurrent());
-  rtc::scoped_refptr<RTCStatsReport> report = RTCStatsReport::Create();
+  rtc::scoped_refptr<RTCStatsReport> report = RTCStatsReport::Create(
+      timestamp_us);
 
   SessionStats session_stats;
   if (pc_->session()->GetTransportStats(&session_stats)) {
     std::map<std::string, CertificateStatsPair> transport_cert_stats =
         PrepareTransportCertificateStats_s(session_stats);
 
     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());
   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();
+  rtc::scoped_refptr<RTCStatsReport> report = RTCStatsReport::Create(
+      timestamp_us);
 
   // TODO(hbos): Gather stats on worker thread.
+  // pc_->session()'s channels are owned by the signaling thread but there are
+  // some stats that are gathered on the worker thread. Instead of a synchronous
+  // invoke on "s->w" we could to the "w" work here asynchronously if it wasn't
+  // for the ownership issue. Synchronous invokes in other places makes it
+  // difficult to introduce locks without introducing deadlocks and the channels
+  // are not reference counted.
 
   AddPartialResults(report);
 }
 
 void RTCStatsCollector::ProducePartialResultsOnNetworkThread(
     int64_t timestamp_us) {
   RTC_DCHECK(network_thread_->IsCurrent());
-  rtc::scoped_refptr<RTCStatsReport> report = RTCStatsReport::Create();
+  rtc::scoped_refptr<RTCStatsReport> report = RTCStatsReport::Create(
+      timestamp_us);
 
   // TODO(hbos): Gather stats on network thread.
+  // pc_->session()'s channels are owned by the signaling thread but there are
+  // some stats that are gathered on the network thread. Instead of a
+  // synchronous invoke on "s->n" we could to the "n" work here asynchronously
+  // if it wasn't for the ownership issue. Synchronous invokes in other places
+  // makes it difficult to introduce locks without introducing deadlocks and the
+  // channels are not reference counted.
 
   AddPartialResults(report);
 }
 
 void RTCStatsCollector::AddPartialResults(
     const rtc::scoped_refptr<RTCStatsReport>& partial_report) {
   if (!signaling_thread_->IsCurrent()) {
     invoker_.AsyncInvoke<void>(RTC_FROM_HERE, signaling_thread_,
         rtc::Bind(&RTCStatsCollector::AddPartialResults_s,
                   rtc::scoped_refptr<RTCStatsCollector>(this),
@@ skipping 116 matching lines @@
         // crbug.com/633550
         candidate_pair_stats->writable = info.writable;
         candidate_pair_stats->bytes_sent =
             static_cast<uint64_t>(info.sent_total_bytes);
         candidate_pair_stats->bytes_received =
             static_cast<uint64_t>(info.recv_total_bytes);
         // TODO(hbos): The |info.rtt| measurement is smoothed. It shouldn't be
         // smoothed according to the spec. crbug.com/633550. See
         // https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-currentrtt
         candidate_pair_stats->current_rtt =
-            static_cast<double>(info.rtt) / 1000.0;
+            static_cast<double>(info.rtt) / rtc::kNumMillisecsPerSec;
         candidate_pair_stats->requests_sent =
             static_cast<uint64_t>(info.sent_ping_requests_total);
         candidate_pair_stats->responses_received =
             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));
       }
     }
@@ skipping 43 matching lines @@
   // There is always just one |RTCPeerConnectionStats| so its |id| can be a
   // constant.
   std::unique_ptr<RTCPeerConnectionStats> stats(
     new RTCPeerConnectionStats("RTCPeerConnection", timestamp_us));
   stats->data_channels_opened = data_channels_opened;
   stats->data_channels_closed = static_cast<uint32_t>(data_channels.size()) -
                                 data_channels_opened;
   report->AddStats(std::move(stats));
 }
 
+void RTCStatsCollector::ProduceRTPStreamStats_s(
+    int64_t timestamp_us, const SessionStats& session_stats,
+    RTCStatsReport* report) const {
+  RTC_DCHECK(signaling_thread_->IsCurrent());
+
+  // Audio
+  if (pc_->session()->voice_channel()) {
+    cricket::VoiceMediaInfo voice_media_info;
+    if (pc_->session()->voice_channel()->GetStats(&voice_media_info)) {
+      std::string transport_id = RTCTransportStatsIDFromBaseChannel(
+          session_stats.proxy_to_transport, *pc_->session()->voice_channel());
+      for (const cricket::VoiceSenderInfo& voice_sender_info :
+           voice_media_info.senders) {
+        // TODO(nisse): SSRC == 0 currently means none. Delete check when that
+        // is fixed.
+        if (voice_sender_info.ssrc() == 0)
+          continue;
+        std::unique_ptr<RTCOutboundRTPStreamStats> outbound_audio(
+            new RTCOutboundRTPStreamStats(
+                RTCOutboundRTPStreamStatsIDFromSSRC(
+                    true, voice_sender_info.ssrc()),
+                timestamp_us));
+        SetOutboundRTPStreamStatsFromVoiceSenderInfo(
+            voice_sender_info, outbound_audio.get());
+        if (!transport_id.empty())
+          outbound_audio->transport_id = transport_id;
+        report->AddStats(std::move(outbound_audio));
+      }
+    }
+  }
+  // Video
+  if (pc_->session()->video_channel()) {
+    cricket::VideoMediaInfo video_media_info;
+    if (pc_->session()->video_channel()->GetStats(&video_media_info)) {
+      std::string transport_id = RTCTransportStatsIDFromBaseChannel(
+          session_stats.proxy_to_transport, *pc_->session()->video_channel());
+      for (const cricket::VideoSenderInfo& video_sender_info :
+           video_media_info.senders) {
+        // TODO(nisse): SSRC == 0 currently means none. Delete check when that
+        // is fixed.
+        if (video_sender_info.ssrc() == 0)
+          continue;
+        std::unique_ptr<RTCOutboundRTPStreamStats> outbound_video(
+            new RTCOutboundRTPStreamStats(
+                RTCOutboundRTPStreamStatsIDFromSSRC(
+                    false, video_sender_info.ssrc()),
+                timestamp_us));
+        SetOutboundRTPStreamStatsFromVideoSenderInfo(
+            video_sender_info, outbound_video.get());
+        if (!transport_id.empty())
+          outbound_video->transport_id = transport_id;
+        report->AddStats(std::move(outbound_video));
+      }
+    }
+  }
+}
+
 void RTCStatsCollector::ProduceTransportStats_s(
     int64_t timestamp_us, const SessionStats& session_stats,
     const std::map<std::string, CertificateStatsPair>& transport_cert_stats,
     RTCStatsReport* report) const {
   RTC_DCHECK(signaling_thread_->IsCurrent());
   for (const auto& transport : session_stats.transport_stats) {
     // Get reference to RTCP channel, if it exists.
     std::string rtcp_transport_stats_id;
     for (const auto& channel_stats : transport.second.channel_stats) {
       if (channel_stats.component ==
@@ skipping 83 matching lines @@
     const std::string& type) {
   return CandidateTypeToRTCIceCandidateType(type);
 }
 
 const char* DataStateToRTCDataChannelStateForTesting(
     DataChannelInterface::DataState state) {
   return DataStateToRTCDataChannelState(state);
 }
 
 }  // namespace webrtc