RTCInboundRTPStreamStats 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.
  */
 
@@ skipping 35 matching lines @@
 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 RTCInboundRTPStreamStatsIDFromSSRC(bool audio, uint32_t ssrc) {
+  return audio ? "RTCInboundRTPAudioStream_" + rtc::ToString<>(ssrc)
+               : "RTCInboundRTPVideoStream_" + rtc::ToString<>(ssrc);
+}
+
 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;
@@ skipping 15 matching lines @@
     case DataChannelInterface::kClosing:
       return RTCDataChannelState::kClosing;
     case DataChannelInterface::kClosed:
       return RTCDataChannelState::kClosed;
     default:
       RTC_NOTREACHED();
       return nullptr;
   }
 }
 
+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 =
+      static_cast<uint32_t>(media_receiver_info.packets_rcvd);
+  inbound_stats->bytes_received =
+      static_cast<uint64_t>(media_receiver_info.bytes_rcvd);
+  inbound_stats->fraction_lost =
+      static_cast<double>(media_receiver_info.fraction_lost);
+}
+
+void SetInboundRTPStreamStatsFromVoiceReceiverInfo(
+    const cricket::VoiceReceiverInfo& voice_receiver_info,
+    RTCInboundRTPStreamStats* inbound_stats) {
+  SetInboundRTPStreamStatsFromMediaReceiverInfo(
+      voice_receiver_info, inbound_stats);
+  inbound_stats->media_type = "audio";
+  inbound_stats->jitter =
+      static_cast<double>(voice_receiver_info.jitter_buffer_ms) /

[Taylor Brandstetter, 2016/10/27 21:26:03]

I don't think this is right; it looks like jitter_buffer_ms is the delay used by
the jitter buffer, rather than the raw jitter. I think we want "jitter_ms". Same
below.

[hbos, 2016/10/28 10:03:48]

Done. The audio case has jitter_ms, jitter_buffer_ms and
jitter_buffer_preferred_ms but the video case only has jitter_buffer_ms (with
comment "Jitter (network-related) latency").
I updated to use jitter_ms in the audio case and skipped the stat for the video
case.

+      rtc::kNumMillisecsPerSec;
+}
+
+void SetInboundRTPStreamStatsFromVideoReceiverInfo(
+    const cricket::VideoReceiverInfo& video_receiver_info,
+    RTCInboundRTPStreamStats* inbound_stats) {
+  SetInboundRTPStreamStatsFromMediaReceiverInfo(
+      video_receiver_info, inbound_stats);
+  inbound_stats->media_type = "video";
+  inbound_stats->jitter =
+      static_cast<double>(video_receiver_info.jitter_buffer_ms) /
+      rtc::kNumMillisecsPerSec;
+}
+
 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 =
@@ skipping 356 matching lines @@
     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());
+      // Inbound
+      for (const cricket::VoiceReceiverInfo& voice_receiver_info :
+           voice_media_info.receivers) {
+        // TODO(nisse): SSRC == 0 currently means none. Delete check when that
+        // is fixed.
+        if (voice_receiver_info.ssrc() == 0)
+          continue;
+        std::unique_ptr<RTCInboundRTPStreamStats> inbound_audio(
+            new RTCInboundRTPStreamStats(
+                RTCInboundRTPStreamStatsIDFromSSRC(
+                    true, voice_receiver_info.ssrc()),
+                timestamp_us));
+        SetInboundRTPStreamStatsFromVoiceReceiverInfo(
+            voice_receiver_info, inbound_audio.get());
+        if (!transport_id.empty())

[Taylor Brandstetter, 2016/10/27 21:26:03]

Is it ever possible for transport_id to be empty here?

[Taylor Brandstetter, 2016/10/27 21:28:30]

Oh, I see now. It's possible given how "proxy_to_transport" works, but it's one
of those things that "shouldn't" ever happen. So I'd give it an RTC_DCHECK at
least.

[hbos, 2016/10/28 10:03:48]

Done.

+          inbound_audio->transport_id = transport_id;
+        report->AddStats(std::move(inbound_audio));
+      }
+      // Outbound
       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());
+      // Inbound
+      for (const cricket::VideoReceiverInfo& video_receiver_info :
+           video_media_info.receivers) {
+        // TODO(nisse): SSRC == 0 currently means none. Delete check when that
+        // is fixed.
+        if (video_receiver_info.ssrc() == 0)
+          continue;
+        std::unique_ptr<RTCInboundRTPStreamStats> inbound_video(
+            new RTCInboundRTPStreamStats(
+                RTCInboundRTPStreamStatsIDFromSSRC(
+                    false, video_receiver_info.ssrc()),
+                timestamp_us));
+        SetInboundRTPStreamStatsFromVideoReceiverInfo(
+            video_receiver_info, inbound_video.get());
+        if (!transport_id.empty())
+          inbound_video->transport_id = transport_id;
+        report->AddStats(std::move(inbound_video));
+      }
+      // Outbound
       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()),
@@ skipping 104 matching lines @@
     const std::string& type) {
   return CandidateTypeToRTCIceCandidateType(type);
 }
 
 const char* DataStateToRTCDataChannelStateForTesting(
     DataChannelInterface::DataState state) {
   return DataStateToRTCDataChannelState(state);
 }
 
 }  // namespace webrtc