RTCInboundRTPStreamStats[1] added.

Not all stats are collected in this CL, this must be addressed before closing the issue. [1] https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict* Re-landed after having to be reverted https://codereview.webrtc.org/2470683002/ due to depending on a CL that was reverted. Now that that has re-landed https://codereview.webrtc.org/2470703002/ this is ready to re-land. BUG=chromium:627816, chromium:657855, chromium:657854 R=hta@webrtc.org TBR=deadbeef@webrtc.org Review-Url: https://codereview.webrtc.org/2465173003 Cr-Commit-Position: refs/heads/master@{#14868}
2016-11-01 03:00:17 -07:00
parent b84ad63b0a
commit eeafe94f28
5 changed files with 306 additions and 21 deletions
--- a/webrtc/api/rtcstatscollector.cc
+++ b/webrtc/api/rtcstatscollector.cc
@ -53,6 +53,11 @@ std::string RTCTransportStatsIDFromBaseChannel(
      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);
@ -88,6 +93,42 @@ const char* DataStateToRTCDataChannelState(
  }
 }

+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_ms) /
+          rtc::kNumMillisecsPerSec;
+}
+
+void SetInboundRTPStreamStatsFromVideoReceiverInfo(
+    const cricket::VideoReceiverInfo& video_receiver_info,
+    RTCInboundRTPStreamStats* inbound_stats) {
+  SetInboundRTPStreamStatsFromMediaReceiverInfo(
+      video_receiver_info, inbound_stats);
+  inbound_stats->media_type = "video";
+}
+
 void SetOutboundRTPStreamStatsFromMediaSenderInfo(
    const cricket::MediaSenderInfo& media_sender_info,
    RTCOutboundRTPStreamStats* outbound_stats) {
@ -460,6 +501,25 @@ void RTCStatsCollector::ProduceRTPStreamStats_s(
    if (pc_->session()->voice_channel()->GetStats(&voice_media_info)) {
      std::string transport_id = RTCTransportStatsIDFromBaseChannel(
          session_stats.proxy_to_transport, *pc_->session()->voice_channel());
+      RTC_DCHECK(!transport_id.empty());
+      // 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());
+        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
@ -473,8 +533,7 @@ void RTCStatsCollector::ProduceRTPStreamStats_s(
                timestamp_us));
        SetOutboundRTPStreamStatsFromVoiceSenderInfo(
            voice_sender_info, outbound_audio.get());
-        if (!transport_id.empty())
-          outbound_audio->transport_id = transport_id;
+        outbound_audio->transport_id = transport_id;
        report->AddStats(std::move(outbound_audio));
      }
    }
@ -485,6 +544,25 @@ void RTCStatsCollector::ProduceRTPStreamStats_s(
    if (pc_->session()->video_channel()->GetStats(&video_media_info)) {
      std::string transport_id = RTCTransportStatsIDFromBaseChannel(
          session_stats.proxy_to_transport, *pc_->session()->video_channel());
+      RTC_DCHECK(!transport_id.empty());
+      // 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());
+        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
@ -498,8 +576,7 @@ void RTCStatsCollector::ProduceRTPStreamStats_s(
                timestamp_us));
        SetOutboundRTPStreamStatsFromVideoSenderInfo(
            video_sender_info, outbound_video.get());
-        if (!transport_id.empty())
-          outbound_video->transport_id = transport_id;
+        outbound_video->transport_id = transport_id;
        report->AddStats(std::move(outbound_video));
      }
    }
--- a/webrtc/api/rtcstatscollector.h
+++ b/webrtc/api/rtcstatscollector.h
@ -101,9 +101,7 @@ class RTCStatsCollector : public virtual rtc::RefCountInterface {
  // Produces |RTCPeerConnectionStats|.
  void ProducePeerConnectionStats_s(
      int64_t timestamp_us, RTCStatsReport* report) const;
-  // Produces |RTCOutboundRTPStreamStats|. TODO(hbos): Produce both types of
-  // |RTCRTPStreamStats|, the other one being |RTCInboundRTPStreamStats|.
-  // crbug.com/657855
+  // Produces |RTCInboundRTPStreamStats| and |RTCOutboundRTPStreamStats|.
  void ProduceRTPStreamStats_s(
      int64_t timestamp_us, const SessionStats& session_stats,
      RTCStatsReport* report) const;
--- a/webrtc/api/rtcstatscollector_unittest.cc
+++ b/webrtc/api/rtcstatscollector_unittest.cc
@ -70,6 +70,10 @@ void PrintTo(const RTCPeerConnectionStats& stats, ::std::ostream* os) {
  *os << stats.ToString();
 }

+void PrintTo(const RTCInboundRTPStreamStats& stats, ::std::ostream* os) {
+  *os << stats.ToString();
+}
+
 void PrintTo(const RTCOutboundRTPStreamStats& stats, ::std::ostream* os) {
  *os << stats.ToString();
 }
@ -975,6 +979,116 @@ TEST_F(RTCStatsCollectorTest, CollectRTCPeerConnectionStats) {
  }
 }

+TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Audio) {
+  MockVoiceMediaChannel* voice_media_channel = new MockVoiceMediaChannel();
+  cricket::VoiceChannel voice_channel(
+      test_->worker_thread(), test_->network_thread(), test_->media_engine(),
+      voice_media_channel, nullptr, "VoiceContentName", false);
+
+  cricket::VoiceMediaInfo voice_media_info;
+  voice_media_info.receivers.push_back(cricket::VoiceReceiverInfo());
+  voice_media_info.receivers[0].local_stats.push_back(
+      cricket::SsrcReceiverInfo());
+  voice_media_info.receivers[0].local_stats[0].ssrc = 1;
+  voice_media_info.receivers[0].packets_rcvd = 2;
+  voice_media_info.receivers[0].bytes_rcvd = 3;
+  voice_media_info.receivers[0].jitter_ms = 4500;
+  voice_media_info.receivers[0].fraction_lost = 5.5f;
+  EXPECT_CALL(*voice_media_channel, GetStats(_))
+      .WillOnce(DoAll(SetArgPointee<0>(voice_media_info), Return(true)));
+
+  SessionStats session_stats;
+  session_stats.proxy_to_transport["VoiceContentName"] = "TransportName";
+  session_stats.transport_stats["TransportName"].transport_name =
+      "TransportName";
+
+  // Make sure the associated |RTCTransportStats| is created.
+  cricket::TransportChannelStats channel_stats;
+  channel_stats.component = cricket::ICE_CANDIDATE_COMPONENT_RTP;
+  session_stats.transport_stats["TransportName"].channel_stats.push_back(
+      channel_stats);
+
+  EXPECT_CALL(test_->session(), GetTransportStats(_))
+      .WillRepeatedly(DoAll(SetArgPointee<0>(session_stats), Return(true)));
+  EXPECT_CALL(test_->session(), voice_channel())
+      .WillRepeatedly(Return(&voice_channel));
+
+  rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
+
+  RTCInboundRTPStreamStats expected_audio(
+      "RTCInboundRTPAudioStream_1", report->timestamp_us());
+  expected_audio.ssrc = "1";
+  expected_audio.is_remote = false;
+  expected_audio.media_type = "audio";
+  expected_audio.transport_id = "RTCTransport_TransportName_" +
+      rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP);
+  expected_audio.packets_received = 2;
+  expected_audio.bytes_received = 3;
+  expected_audio.jitter = 4.5;
+  expected_audio.fraction_lost = 5.5;
+
+  ASSERT(report->Get(expected_audio.id()));
+  const RTCInboundRTPStreamStats& audio = report->Get(
+      expected_audio.id())->cast_to<RTCInboundRTPStreamStats>();
+  EXPECT_EQ(audio, expected_audio);
+
+  EXPECT_TRUE(report->Get(*expected_audio.transport_id));
+}
+
+TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
+  MockVideoMediaChannel* video_media_channel = new MockVideoMediaChannel();
+  cricket::VideoChannel video_channel(
+      test_->worker_thread(), test_->network_thread(), video_media_channel,
+      nullptr, "VideoContentName", false);
+
+  cricket::VideoMediaInfo video_media_info;
+  video_media_info.receivers.push_back(cricket::VideoReceiverInfo());
+  video_media_info.receivers[0].local_stats.push_back(
+      cricket::SsrcReceiverInfo());
+  video_media_info.receivers[0].local_stats[0].ssrc = 1;
+  video_media_info.receivers[0].packets_rcvd = 2;
+  video_media_info.receivers[0].bytes_rcvd = 3;
+  video_media_info.receivers[0].fraction_lost = 4.5f;
+  EXPECT_CALL(*video_media_channel, GetStats(_))
+      .WillOnce(DoAll(SetArgPointee<0>(video_media_info), Return(true)));
+
+  SessionStats session_stats;
+  session_stats.proxy_to_transport["VideoContentName"] = "TransportName";
+  session_stats.transport_stats["TransportName"].transport_name =
+      "TransportName";
+
+  // Make sure the associated |RTCTransportStats| is created.
+  cricket::TransportChannelStats channel_stats;
+  channel_stats.component = cricket::ICE_CANDIDATE_COMPONENT_RTP;
+  session_stats.transport_stats["TransportName"].channel_stats.push_back(
+      channel_stats);
+
+  EXPECT_CALL(test_->session(), GetTransportStats(_))
+      .WillRepeatedly(DoAll(SetArgPointee<0>(session_stats), Return(true)));
+  EXPECT_CALL(test_->session(), video_channel())
+      .WillRepeatedly(Return(&video_channel));
+
+  rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
+
+  RTCInboundRTPStreamStats expected_audio(
+      "RTCInboundRTPVideoStream_1", report->timestamp_us());
+  expected_audio.ssrc = "1";
+  expected_audio.is_remote = false;
+  expected_audio.media_type = "video";
+  expected_audio.transport_id = "RTCTransport_TransportName_" +
+      rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP);
+  expected_audio.packets_received = 2;
+  expected_audio.bytes_received = 3;
+  expected_audio.fraction_lost = 4.5;
+
+  ASSERT(report->Get(expected_audio.id()));
+  const RTCInboundRTPStreamStats& audio = report->Get(
+      expected_audio.id())->cast_to<RTCInboundRTPStreamStats>();
+  EXPECT_EQ(audio, expected_audio);
+
+  EXPECT_TRUE(report->Get(*expected_audio.transport_id));
+}
+
 TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) {
  MockVoiceMediaChannel* voice_media_channel = new MockVoiceMediaChannel();
  cricket::VoiceChannel voice_channel(
@ -987,7 +1101,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) {
  voice_media_info.senders[0].local_stats[0].ssrc = 1;
  voice_media_info.senders[0].packets_sent = 2;
  voice_media_info.senders[0].bytes_sent = 3;
-  voice_media_info.senders[0].rtt_ms = 4500.0;
+  voice_media_info.senders[0].rtt_ms = 4500;
  EXPECT_CALL(*voice_media_channel, GetStats(_))
      .WillOnce(DoAll(SetArgPointee<0>(voice_media_info), Return(true)));

@ -999,12 +1113,6 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) {
  // Make sure the associated |RTCTransportStats| is created.
  cricket::TransportChannelStats channel_stats;
  channel_stats.component = cricket::ICE_CANDIDATE_COMPONENT_RTP;
-  cricket::ConnectionInfo connection_info;
-  connection_info.local_candidate = *CreateFakeCandidate(
-      "42.42.42.42", 42, "protocol", cricket::LOCAL_PORT_TYPE, 42).get();
-  connection_info.remote_candidate = *CreateFakeCandidate(
-      "42.42.42.42", 42, "protocol", cricket::LOCAL_PORT_TYPE, 42).get();
-  channel_stats.connection_infos.push_back(connection_info);
  session_stats.transport_stats["TransportName"].channel_stats.push_back(
      channel_stats);

@ -1049,7 +1157,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) {
  video_media_info.senders[0].nacks_rcvd = 4;
  video_media_info.senders[0].packets_sent = 5;
  video_media_info.senders[0].bytes_sent = 6;
-  video_media_info.senders[0].rtt_ms = 7500.0;
+  video_media_info.senders[0].rtt_ms = 7500;
  EXPECT_CALL(*video_media_channel, GetStats(_))
      .WillOnce(DoAll(SetArgPointee<0>(video_media_info), Return(true)));

@ -1061,12 +1169,6 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) {
  // Make sure the associated |RTCTransportStats| is created.
  cricket::TransportChannelStats channel_stats;
  channel_stats.component = cricket::ICE_CANDIDATE_COMPONENT_RTP;
-  cricket::ConnectionInfo connection_info;
-  connection_info.local_candidate = *CreateFakeCandidate(
-      "42.42.42.42", 42, "protocol", cricket::LOCAL_PORT_TYPE, 42).get();
-  connection_info.remote_candidate = *CreateFakeCandidate(
-      "42.42.42.42", 42, "protocol", cricket::LOCAL_PORT_TYPE, 42).get();
-  channel_stats.connection_infos.push_back(connection_info);
  session_stats.transport_stats["TransportName"].channel_stats.push_back(
      channel_stats);

--- a/webrtc/api/stats/rtcstats_objects.h
+++ b/webrtc/api/stats/rtcstats_objects.h
@ -239,6 +239,48 @@ class RTCRTPStreamStats : public RTCStats {
  RTCRTPStreamStats(std::string&& id, int64_t timestamp_us);
 };

+// https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*
+// TODO(hbos): Finish implementation and support the remote case
+// |is_remote = true|. Tracking bug crbug.com/657855
+class RTCInboundRTPStreamStats final : public RTCRTPStreamStats {
+ public:
+  WEBRTC_RTCSTATS_DECL();
+
+  RTCInboundRTPStreamStats(const std::string& id, int64_t timestamp_us);
+  RTCInboundRTPStreamStats(std::string&& id, int64_t timestamp_us);
+  RTCInboundRTPStreamStats(const RTCInboundRTPStreamStats& other);
+  ~RTCInboundRTPStreamStats() override;
+
+  RTCStatsMember<uint32_t> packets_received;
+  RTCStatsMember<uint64_t> bytes_received;
+  // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+  RTCStatsMember<uint32_t> packets_lost;
+  // TODO(hbos): Not collected in the "video" case by |RTCStatsCollector|.
+  // crbug.com/657855
+  RTCStatsMember<double> jitter;
+  RTCStatsMember<double> fraction_lost;
+  // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+  RTCStatsMember<uint32_t> packets_discarded;
+  // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+  RTCStatsMember<uint32_t> packets_repaired;
+  // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+  RTCStatsMember<uint32_t> burst_packets_lost;
+  // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+  RTCStatsMember<uint32_t> burst_packets_discarded;
+  // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+  RTCStatsMember<uint32_t> burst_loss_count;
+  // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+  RTCStatsMember<uint32_t> burst_discard_count;
+  // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+  RTCStatsMember<double> burst_loss_rate;
+  // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+  RTCStatsMember<double> burst_discard_rate;
+  // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+  RTCStatsMember<double> gap_loss_rate;
+  // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657855
+  RTCStatsMember<double> gap_discard_rate;
+};
+
 // https://w3c.github.io/webrtc-stats/#outboundrtpstats-dict*
 // TODO(hbos): Finish implementation and support the remote case
 // |is_remote = true|. Tracking bug crbug.com/657856
--- a/webrtc/stats/rtcstats_objects.cc
+++ b/webrtc/stats/rtcstats_objects.cc
@ -346,6 +346,72 @@ RTCRTPStreamStats::RTCRTPStreamStats(
 RTCRTPStreamStats::~RTCRTPStreamStats() {
 }

+WEBRTC_RTCSTATS_IMPL(
+    RTCInboundRTPStreamStats, RTCRTPStreamStats, "inbound-rtp",
+    &packets_received,
+    &bytes_received,
+    &packets_lost,
+    &jitter,
+    &fraction_lost,
+    &packets_discarded,
+    &packets_repaired,
+    &burst_packets_lost,
+    &burst_packets_discarded,
+    &burst_loss_count,
+    &burst_discard_count,
+    &burst_loss_rate,
+    &burst_discard_rate,
+    &gap_loss_rate,
+    &gap_discard_rate);
+
+RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(
+    const std::string& id, int64_t timestamp_us)
+    : RTCInboundRTPStreamStats(std::string(id), timestamp_us) {
+}
+
+RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(
+    std::string&& id, int64_t timestamp_us)
+    : RTCRTPStreamStats(std::move(id), timestamp_us),
+      packets_received("packetsReceived"),
+      bytes_received("bytesReceived"),
+      packets_lost("packetsLost"),
+      jitter("jitter"),
+      fraction_lost("fractionLost"),
+      packets_discarded("packetsDiscarded"),
+      packets_repaired("packetsRepaired"),
+      burst_packets_lost("burstPacketsLost"),
+      burst_packets_discarded("burstPacketsDiscarded"),
+      burst_loss_count("burstLossCount"),
+      burst_discard_count("burstDiscardCount"),
+      burst_loss_rate("burstLossRate"),
+      burst_discard_rate("burstDiscardRate"),
+      gap_loss_rate("gapLossRate"),
+      gap_discard_rate("gapDiscardRate") {
+}
+
+RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(
+    const RTCInboundRTPStreamStats& other)
+    : RTCRTPStreamStats(other),
+      packets_received(other.packets_received),
+      bytes_received(other.bytes_received),
+      packets_lost(other.packets_lost),
+      jitter(other.jitter),
+      fraction_lost(other.fraction_lost),
+      packets_discarded(other.packets_discarded),
+      packets_repaired(other.packets_repaired),
+      burst_packets_lost(other.burst_packets_lost),
+      burst_packets_discarded(other.burst_packets_discarded),
+      burst_loss_count(other.burst_loss_count),
+      burst_discard_count(other.burst_discard_count),
+      burst_loss_rate(other.burst_loss_rate),
+      burst_discard_rate(other.burst_discard_rate),
+      gap_loss_rate(other.gap_loss_rate),
+      gap_discard_rate(other.gap_discard_rate) {
+}
+
+RTCInboundRTPStreamStats::~RTCInboundRTPStreamStats() {
+}
+
 WEBRTC_RTCSTATS_IMPL(
    RTCOutboundRTPStreamStats, RTCRTPStreamStats, "outbound-rtp",
    &packets_sent,