Bug 1207824: Add Telemetry for WebRTC call type, simultaneous tracks, and renegotiations r=bwc

media/webrtc/signaling/src/jsep/JsepSession.h

@@ -15,13 +15,13 @@
 #include "signaling/src/jsep/JsepTransport.h"
 #include "signaling/src/sdp/Sdp.h"
 
+#include "JsepTrack.h"
 
 namespace mozilla {
 
 // Forward declarations
 class JsepCodecDescription;
 class JsepTrack;
-struct JsepTrackPair;
 
 enum JsepSignalingState {
   kJsepStateStable,
@@ -57,7 +57,7 @@ class JsepSession
 {
 public:
   explicit JsepSession(const std::string& name)
-      : mName(name), mState(kJsepStateStable)
+    : mName(name), mState(kJsepStateStable), mNegotiations(0)
   {
   }
   virtual ~JsepSession() {}
@@ -75,6 +75,11 @@ public:
   {
     return mState;
   }
+  virtual uint32_t
+  GetNegotiations() const
+  {
+    return mNegotiations;
+  }
 
   // Set up the ICE And DTLS data.
   virtual nsresult SetIceCredentials(const std::string& ufrag,
@@ -167,9 +172,30 @@ public:
 
   virtual bool AllLocalTracksAreAssigned() const = 0;
 
+  void
+  CountTracks(uint16_t (&receiving)[SdpMediaSection::kMediaTypes],
+              uint16_t (&sending)[SdpMediaSection::kMediaTypes]) const
+  {
+    auto trackPairs = GetNegotiatedTrackPairs();
+
+    memset(receiving, 0, sizeof(receiving));
+    memset(sending, 0, sizeof(sending));
+
+    for (auto& pair : trackPairs) {
+      if (pair.mReceiving) {
+        receiving[pair.mReceiving->GetMediaType()]++;
+      }
+
+      if (pair.mSending) {
+        sending[pair.mSending->GetMediaType()]++;
+      }
+    }
+  }
+
 protected:
   const std::string mName;
   JsepSignalingState mState;
+  uint32_t mNegotiations;
 };
 
 } // namespace mozilla

media/webrtc/signaling/src/jsep/JsepSessionImpl.cpp

@@ -1299,6 +1299,8 @@ JsepSessionImpl::HandleNegotiatedSession(const UniquePtr<Sdp>& local,
   mNegotiatedTrackPairs = trackPairs;
 
   mGeneratedLocalDescription.reset();
+
+  mNegotiations++;
   return NS_OK;
 }
 

media/webrtc/signaling/src/peerconnection/PeerConnectionImpl.cpp

@@ -410,6 +410,8 @@ PeerConnectionImpl::PeerConnectionImpl(const GlobalObject* aGlobal)
   mAllowIceLinkLocal = Preferences::GetBool(
     "media.peerconnection.ice.link_local", false);
 #endif
+  memset(mMaxReceiving, 0, sizeof(mMaxReceiving));
+  memset(mMaxSending, 0, sizeof(mMaxSending));
 }
 
 PeerConnectionImpl::~PeerConnectionImpl()
@@ -2486,6 +2488,57 @@ PeerConnectionImpl::PluginCrash(uint32_t aPluginID,
   return true;
 }
 
+void
+PeerConnectionImpl::RecordEndOfCallTelemetry() const
+{
+#if !defined(MOZILLA_EXTERNAL_LINKAGE)
+  // Bitmask used for WEBRTC/LOOP_CALL_TYPE telemetry reporting
+  static const uint32_t kAudioTypeMask = 1;
+  static const uint32_t kVideoTypeMask = 2;
+  static const uint32_t kDataChannelTypeMask = 4;
+
+  // Report end-of-call Telemetry
+  if (mJsepSession->GetNegotiations() > 0) {
+    Telemetry::Accumulate(mIsLoop ? Telemetry::LOOP_RENEGOTIATIONS :
+                          Telemetry::WEBRTC_RENEGOTIATIONS,
+                          mJsepSession->GetNegotiations()-1);
+  }
+  Telemetry::Accumulate(mIsLoop ? Telemetry::LOOP_MAX_VIDEO_SEND_TRACK :
+                        Telemetry::WEBRTC_MAX_VIDEO_SEND_TRACK,
+                        mMaxSending[SdpMediaSection::MediaType::kVideo]);
+  Telemetry::Accumulate(mIsLoop ? Telemetry::LOOP_MAX_VIDEO_RECEIVE_TRACK :
+                        Telemetry::WEBRTC_MAX_VIDEO_RECEIVE_TRACK,
+                        mMaxReceiving[SdpMediaSection::MediaType::kVideo]);
+  Telemetry::Accumulate(mIsLoop ? Telemetry::LOOP_MAX_AUDIO_SEND_TRACK :
+                        Telemetry::WEBRTC_MAX_AUDIO_SEND_TRACK,
+                        mMaxSending[SdpMediaSection::MediaType::kAudio]);
+  Telemetry::Accumulate(mIsLoop ? Telemetry::LOOP_MAX_AUDIO_RECEIVE_TRACK :
+                        Telemetry::WEBRTC_MAX_AUDIO_RECEIVE_TRACK,
+                        mMaxReceiving[SdpMediaSection::MediaType::kAudio]);
+  // DataChannels appear in both Sending and Receiving
+  Telemetry::Accumulate(mIsLoop ? Telemetry::LOOP_DATACHANNEL_NEGOTIATED :
+                        Telemetry::WEBRTC_DATACHANNEL_NEGOTIATED,
+                        mMaxSending[SdpMediaSection::MediaType::kApplication]);
+  // Enumerated/bitmask: 1 = Audio, 2 = Video, 4 = DataChannel
+  // A/V = 3, A/V/D = 7, etc
+  uint32_t type = 0;
+  if (mMaxSending[SdpMediaSection::MediaType::kAudio] ||
+      mMaxReceiving[SdpMediaSection::MediaType::kAudio]) {
+    type = kAudioTypeMask;
+  }
+  if (mMaxSending[SdpMediaSection::MediaType::kVideo] ||
+      mMaxReceiving[SdpMediaSection::MediaType::kVideo]) {
+    type |= kVideoTypeMask;
+  }
+  if (mMaxSending[SdpMediaSection::MediaType::kApplication]) {
+    type |= kDataChannelTypeMask;
+  }
+  Telemetry::Accumulate(mIsLoop ? Telemetry::LOOP_CALL_TYPE :
+                        Telemetry::WEBRTC_CALL_TYPE,
+                        type);
+#endif
+}
+
 nsresult
 PeerConnectionImpl::CloseInt()
 {
@@ -2496,6 +2549,7 @@ PeerConnectionImpl::CloseInt()
   // transitioned to connected. As a bonus, this allows us to detect race
   // conditions where a stats dispatch happens right as the PC closes.
   RecordLongtermICEStatistics();
+  RecordEndOfCallTelemetry();
   CSFLogInfo(logTag, "%s: Closing PeerConnectionImpl %s; "
              "ending call", __FUNCTION__, mHandle.c_str());
   if (mJsepSession) {
@@ -2585,6 +2639,23 @@ PeerConnectionImpl::SetSignalingState_m(PCImplSignalingState aSignalingState,
         fireNegotiationNeeded = true;
       }
     }
+
+    // Telemetry: record info on the current state of streams/renegotiations/etc
+    // Note: this code gets run on rollbacks as well!
+
+    // Update the max channels used with each direction for each type
+    uint16_t receiving[SdpMediaSection::kMediaTypes];
+    uint16_t sending[SdpMediaSection::kMediaTypes];
+    mJsepSession->CountTracks(receiving, sending);
+    for (size_t i = 0; i < SdpMediaSection::kMediaTypes; i++) {
+      if (mMaxReceiving[i] < receiving[i]) {
+        mMaxReceiving[i] = receiving[i];
+      }
+      if (mMaxSending[i] < sending[i]) {
+        mMaxSending[i] = sending[i];
+      }
+    }
+
   } else {
     mShouldSuppressNegotiationNeeded = true;
   }

media/webrtc/signaling/src/peerconnection/PeerConnectionImpl.h

@@ -550,6 +550,8 @@ public:
   bool PluginCrash(uint32_t aPluginID,
                    const nsAString& aPluginName);
 
+  void RecordEndOfCallTelemetry() const;
+
   nsresult InitializeDataChannel();
 
   NS_IMETHODIMP_TO_ERRORRESULT_RETREF(nsDOMDataChannel,
@@ -796,6 +798,10 @@ private:
 
   bool mShouldSuppressNegotiationNeeded;
 
+  // storage for Telemetry data
+  uint16_t mMaxReceiving[SdpMediaSection::kMediaTypes];
+  uint16_t mMaxSending[SdpMediaSection::kMediaTypes];
+
 public:
   //these are temporary until the DataChannel Listen/Connect API is removed
   unsigned short listenPort;

media/webrtc/signaling/src/sdp/SdpMediaSection.h

@@ -25,6 +25,8 @@ class SdpMediaSection
 {
 public:
   enum MediaType { kAudio, kVideo, kText, kApplication, kMessage };
+  // don't add to enum to avoid warnings about unhandled enum values
+  static const size_t kMediaTypes = static_cast<size_t>(kMessage) + 1;
 
   enum Protocol {
     kRtpAvp,            // RTP/AVP [RFC4566]

toolkit/components/telemetry/Histograms.json

@@ -6541,6 +6541,52 @@
     "n_buckets": "25",
     "description": "Percentage of time spent in the Stressed load state in calls 5-30 seconds."
   },
+  "WEBRTC_RENEGOTIATIONS": {
+    "expires_in_version": "never",
+    "kind": "linear",
+    "high": "21",
+    "n_buckets": "20",
+    "description": "Number of Renegotiations during each call"
+  },
+  "WEBRTC_MAX_VIDEO_SEND_TRACK": {
+    "expires_in_version": "never",
+    "kind": "linear",
+    "high": "10",
+    "n_buckets": "9",
+    "description": "Number of Video tracks sent simultaneously"
+  },
+  "WEBRTC_MAX_VIDEO_RECEIVE_TRACK": {
+    "expires_in_version": "never",
+    "kind": "linear",
+    "high": "20",
+    "n_buckets": "19",
+    "description": "Number of Video tracks received simultaneously"
+  },
+  "WEBRTC_MAX_AUDIO_SEND_TRACK": {
+    "expires_in_version": "never",
+    "kind": "linear",
+    "high": "20",
+    "n_buckets": "19",
+    "description": "Number of Audio tracks sent simultaneously"
+  },
+  "WEBRTC_MAX_AUDIO_RECEIVE_TRACK": {
+    "expires_in_version": "never",
+    "kind": "linear",
+    "high": "30",
+    "n_buckets": "29",
+    "description": "Number of Audio tracks received simultaneously"
+  },
+  "WEBRTC_DATACHANNEL_NEGOTIATED": {
+    "expires_in_version": "never",
+    "kind": "boolean",
+    "description": "Was DataChannels negotiated"
+  },
+  "WEBRTC_CALL_TYPE": {
+    "expires_in_version": "never",
+    "kind": "enumerated",
+    "n_values": "8",
+    "description": "Type of call: (Bitmask) Audio = 1, Video = 2, DataChannels = 4"
+  },
   "DEVTOOLS_DEBUGGER_RDP_LOCAL_TRACERDETACH_MS": {
     "expires_in_version": "never",
     "kind": "exponential",
@@ -9387,6 +9433,52 @@
     "n_values": 8,
     "description": "Type for media in getUserMedia calls (0=Camera, 1=Screen, 2=Application, 3=Window, 4=Browser, 5=Microphone, 6=AudioCapture, 7=Other)"
   },
+  "LOOP_RENEGOTIATIONS": {
+    "expires_in_version": "never",
+    "kind": "linear",
+    "high": "21",
+    "n_buckets": "20",
+    "description": "Number of Renegotiations during each call"
+  },
+  "LOOP_MAX_VIDEO_SEND_TRACK": {
+    "expires_in_version": "never",
+    "kind": "linear",
+    "high": "10",
+    "n_buckets": "9",
+    "description": "Number of Video tracks sent simultaneously"
+  },
+  "LOOP_MAX_VIDEO_RECEIVE_TRACK": {
+    "expires_in_version": "never",
+    "kind": "linear",
+    "high": "20",
+    "n_buckets": "19",
+    "description": "Number of Video tracks received simultaneously"
+  },
+  "LOOP_MAX_AUDIO_SEND_TRACK": {
+    "expires_in_version": "never",
+    "kind": "linear",
+    "high": "20",
+    "n_buckets": "19",
+    "description": "Number of Audio tracks sent simultaneously"
+  },
+  "LOOP_MAX_AUDIO_RECEIVE_TRACK": {
+    "expires_in_version": "never",
+    "kind": "linear",
+    "high": "30",
+    "n_buckets": "29",
+    "description": "Number of Audio tracks received simultaneously"
+  },
+  "LOOP_DATACHANNEL_NEGOTIATED": {
+    "expires_in_version": "never",
+    "kind": "boolean",
+    "description": "Was DataChannels negotiated"
+  },
+  "LOOP_CALL_TYPE": {
+    "expires_in_version": "never",
+    "kind": "enumerated",
+    "n_values": "8",
+    "description": "Type of call: (Bitmask) Audio = 1, Video = 2, DataChannels = 4"
+  },
   "PERF_MONITORING_TEST_CPU_RESCHEDULING_PROPORTION_MOVED": {
     "alert_emails": ["dteller@mozilla.com"],
     "expires_in_version": "45",