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-rw-r--r--media/webrtc/signaling/src/peerconnection/PeerConnectionImpl.cpp109
-rw-r--r--media/webrtc/signaling/src/peerconnection/WebrtcGlobalInformation.cpp16
2 files changed, 4 insertions, 121 deletions
diff --git a/media/webrtc/signaling/src/peerconnection/PeerConnectionImpl.cpp b/media/webrtc/signaling/src/peerconnection/PeerConnectionImpl.cpp
index 33422ed7a..43d10ca86 100644
--- a/media/webrtc/signaling/src/peerconnection/PeerConnectionImpl.cpp
+++ b/media/webrtc/signaling/src/peerconnection/PeerConnectionImpl.cpp
@@ -2245,22 +2245,6 @@ PeerConnectionImpl::AddIceCandidate(const char* aCandidate, const char* aMid, un
CSFLogDebug(logTag, "AddIceCandidate: %s", aCandidate);
-#if !defined(MOZILLA_EXTERNAL_LINKAGE)
- // When remote candidates are added before our ICE ctx is up and running
- // (the transition to New is async through STS, so this is not impossible),
- // we won't record them as trickle candidates. Is this what we want?
- if(!mIceStartTime.IsNull()) {
- TimeDuration timeDelta = TimeStamp::Now() - mIceStartTime;
- if (mIceConnectionState == PCImplIceConnectionState::Failed) {
- Telemetry::Accumulate(Telemetry::WEBRTC_ICE_LATE_TRICKLE_ARRIVAL_TIME,
- timeDelta.ToMilliseconds());
- } else {
- Telemetry::Accumulate(Telemetry::WEBRTC_ICE_ON_TIME_TRICKLE_ARRIVAL_TIME,
- timeDelta.ToMilliseconds());
- }
- }
-#endif
-
nsresult res = mJsepSession->AddRemoteIceCandidate(aCandidate, aMid, aLevel);
if (NS_SUCCEEDED(res)) {
@@ -3011,49 +2995,7 @@ PeerConnectionImpl::PluginCrash(uint32_t aPluginID,
void
PeerConnectionImpl::RecordEndOfCallTelemetry() const
{
- if (!mJsepSession) {
- return;
- }
-
-#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(Telemetry::WEBRTC_RENEGOTIATIONS,
- mJsepSession->GetNegotiations()-1);
- }
- Telemetry::Accumulate(Telemetry::WEBRTC_MAX_VIDEO_SEND_TRACK,
- mMaxSending[SdpMediaSection::MediaType::kVideo]);
- Telemetry::Accumulate(Telemetry::WEBRTC_MAX_VIDEO_RECEIVE_TRACK,
- mMaxReceiving[SdpMediaSection::MediaType::kVideo]);
- Telemetry::Accumulate(Telemetry::WEBRTC_MAX_AUDIO_SEND_TRACK,
- mMaxSending[SdpMediaSection::MediaType::kAudio]);
- Telemetry::Accumulate(Telemetry::WEBRTC_MAX_AUDIO_RECEIVE_TRACK,
- mMaxReceiving[SdpMediaSection::MediaType::kAudio]);
- // DataChannels appear in both Sending and Receiving
- Telemetry::Accumulate(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(Telemetry::WEBRTC_CALL_TYPE,
- type);
-#endif
+ /* STUB */
}
nsresult
@@ -3109,13 +3051,6 @@ PeerConnectionImpl::ShutdownMedia()
pair.second->RemovePrincipalChangeObserver(this);
}
}
-
- // End of call to be recorded in Telemetry
- if (!mStartTime.IsNull()){
- TimeDuration timeDelta = TimeStamp::Now() - mStartTime;
- Telemetry::Accumulate(Telemetry::WEBRTC_CALL_DURATION,
- timeDelta.ToSeconds());
- }
#endif
// Forget the reference so that we can transfer it to
@@ -3423,33 +3358,6 @@ void PeerConnectionImpl::IceConnectionStateChange(
return;
}
-#if !defined(MOZILLA_EXTERNAL_LINKAGE)
- if (!isDone(mIceConnectionState) && isDone(domState)) {
- // mIceStartTime can be null if going directly from New to Closed, in which
- // case we don't count it as a success or a failure.
- if (!mIceStartTime.IsNull()){
- TimeDuration timeDelta = TimeStamp::Now() - mIceStartTime;
- if (isSucceeded(domState)) {
- Telemetry::Accumulate(Telemetry::WEBRTC_ICE_SUCCESS_TIME,
- timeDelta.ToMilliseconds());
- } else if (isFailed(domState)) {
- Telemetry::Accumulate(Telemetry::WEBRTC_ICE_FAILURE_TIME,
- timeDelta.ToMilliseconds());
- }
- }
-
- if (isSucceeded(domState)) {
- Telemetry::Accumulate(
- Telemetry::WEBRTC_ICE_ADD_CANDIDATE_ERRORS_GIVEN_SUCCESS,
- mAddCandidateErrorCount);
- } else if (isFailed(domState)) {
- Telemetry::Accumulate(
- Telemetry::WEBRTC_ICE_ADD_CANDIDATE_ERRORS_GIVEN_FAILURE,
- mAddCandidateErrorCount);
- }
- }
-#endif
-
mIceConnectionState = domState;
if (mIceConnectionState == PCImplIceConnectionState::Connected ||
@@ -3467,10 +3375,6 @@ void PeerConnectionImpl::IceConnectionStateChange(
STAMP_TIMECARD(mTimeCard, "Ice state: new");
break;
case PCImplIceConnectionState::Checking:
-#if !defined(MOZILLA_EXTERNAL_LINKAGE)
- // For telemetry
- mIceStartTime = TimeStamp::Now();
-#endif
STAMP_TIMECARD(mTimeCard, "Ice state: checking");
break;
case PCImplIceConnectionState::Connected:
@@ -4067,16 +3971,7 @@ PeerConnectionImpl::IceStreamReady(NrIceMediaStream *aStream)
//Telemetry for when calls start
void
PeerConnectionImpl::startCallTelem() {
- if (!mStartTime.IsNull()) {
- return;
- }
-
- // Start time for calls
- mStartTime = TimeStamp::Now();
-
- // Increment session call counter
- // If we want to track Loop calls independently here, we need two histograms.
- Telemetry::Accumulate(Telemetry::WEBRTC_CALL_COUNT_2, 1);
+ /* STUB */
}
#endif
diff --git a/media/webrtc/signaling/src/peerconnection/WebrtcGlobalInformation.cpp b/media/webrtc/signaling/src/peerconnection/WebrtcGlobalInformation.cpp
index 96bdd5b70..f283d6111 100644
--- a/media/webrtc/signaling/src/peerconnection/WebrtcGlobalInformation.cpp
+++ b/media/webrtc/signaling/src/peerconnection/WebrtcGlobalInformation.cpp
@@ -1194,18 +1194,8 @@ static void GetStatsForLongTermStorage_s(
rate_limit_bit_pattern |= 2;
}
- if (query->failed) {
- Telemetry::Accumulate(
- Telemetry::WEBRTC_STUN_RATE_LIMIT_EXCEEDED_BY_TYPE_GIVEN_FAILURE,
- rate_limit_bit_pattern);
- } else {
- Telemetry::Accumulate(
- Telemetry::WEBRTC_STUN_RATE_LIMIT_EXCEEDED_BY_TYPE_GIVEN_SUCCESS,
- rate_limit_bit_pattern);
- }
-
- // Even if Telemetry::Accumulate is threadsafe, we still need to send the
- // query back to main, since that is where it must be destroyed.
+ // We still need to send the query back to main, since that is where
+ // it must be destroyed.
NS_DispatchToMainThread(
WrapRunnableNM(
&StoreLongTermICEStatisticsImpl_m,
@@ -1216,8 +1206,6 @@ static void GetStatsForLongTermStorage_s(
void WebrtcGlobalInformation::StoreLongTermICEStatistics(
PeerConnectionImpl& aPc) {
- Telemetry::Accumulate(Telemetry::WEBRTC_ICE_FINAL_CONNECTION_STATE,
- static_cast<uint32_t>(aPc.IceConnectionState()));
if (aPc.IceConnectionState() == PCImplIceConnectionState::New) {
// ICE has not started; we won't have any remote candidates, so recording