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authorMoonchild <mcwerewolf@gmail.com>2018-09-03 18:26:19 +0200
committerGitHub <noreply@github.com>2018-09-03 18:26:19 +0200
commit2e00eb87ef299e6eb7521670e6a6720fee19f5fc (patch)
treec44670a25d942a672951e430499f70978ec7d337 /media
parent45f9a0daad81d1c6a1188b3473e5f0c67d27c0aa (diff)
parentab961aeb54335fd07c66de2e3b8c3b6af6f89ea2 (diff)
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Merge pull request #745 from MoonchildProductions/Kill-Telemetry
Remove all C++ Telemetry Accumulation calls.
Diffstat (limited to 'media')
-rwxr-xr-xmedia/webrtc/signaling/src/media-conduit/AudioConduit.cpp9
-rw-r--r--media/webrtc/signaling/src/media-conduit/CodecStatistics.cpp8
-rw-r--r--media/webrtc/signaling/src/peerconnection/PeerConnectionImpl.cpp109
-rw-r--r--media/webrtc/signaling/src/peerconnection/WebrtcGlobalInformation.cpp16
4 files changed, 4 insertions, 138 deletions
diff --git a/media/webrtc/signaling/src/media-conduit/AudioConduit.cpp b/media/webrtc/signaling/src/media-conduit/AudioConduit.cpp
index 2c57431e7..e36b8b6cf 100755
--- a/media/webrtc/signaling/src/media-conduit/AudioConduit.cpp
+++ b/media/webrtc/signaling/src/media-conduit/AudioConduit.cpp
@@ -706,15 +706,6 @@ WebrtcAudioConduit::GetAudioFrame(int16_t speechData[],
if (GetAVStats(&jitter_buffer_delay_ms,
&playout_buffer_delay_ms,
&avsync_offset_ms)) {
-#if !defined(MOZILLA_EXTERNAL_LINKAGE)
- if (avsync_offset_ms < 0) {
- Telemetry::Accumulate(Telemetry::WEBRTC_AVSYNC_WHEN_VIDEO_LAGS_AUDIO_MS,
- -avsync_offset_ms);
- } else {
- Telemetry::Accumulate(Telemetry::WEBRTC_AVSYNC_WHEN_AUDIO_LAGS_VIDEO_MS,
- avsync_offset_ms);
- }
-#endif
CSFLogError(logTag,
"A/V sync: sync delta: %dms, audio jitter delay %dms, playout delay %dms",
avsync_offset_ms, jitter_buffer_delay_ms, playout_buffer_delay_ms);
diff --git a/media/webrtc/signaling/src/media-conduit/CodecStatistics.cpp b/media/webrtc/signaling/src/media-conduit/CodecStatistics.cpp
index eb03c0bf8..da40a59ea 100644
--- a/media/webrtc/signaling/src/media-conduit/CodecStatistics.cpp
+++ b/media/webrtc/signaling/src/media-conduit/CodecStatistics.cpp
@@ -124,8 +124,6 @@ void VideoCodecStatistics::ReceiveStateChange(const int aChannel,
TimeDuration timeDelta = TimeStamp::Now() - mReceiveFailureTime;
CSFLogError(logTag, "Video error duration: %u ms",
static_cast<uint32_t>(timeDelta.ToMilliseconds()));
- Telemetry::Accumulate(Telemetry::WEBRTC_VIDEO_ERROR_RECOVERY_MS,
- static_cast<uint32_t>(timeDelta.ToMilliseconds()));
mRecoveredLosses++; // to calculate losses per minute
mTotalLossTime += timeDelta; // To calculate % time in recovery
@@ -147,16 +145,10 @@ void VideoCodecStatistics::EndOfCallStats()
if (callDelta.ToSeconds() != 0) {
uint32_t recovered_per_min = mRecoveredBeforeLoss/(callDelta.ToSeconds()/60);
CSFLogError(logTag, "Video recovery before error per min %u", recovered_per_min);
- Telemetry::Accumulate(Telemetry::WEBRTC_VIDEO_RECOVERY_BEFORE_ERROR_PER_MIN,
- recovered_per_min);
uint32_t err_per_min = mRecoveredLosses/(callDelta.ToSeconds()/60);
CSFLogError(logTag, "Video recovery after error per min %u", err_per_min);
- Telemetry::Accumulate(Telemetry::WEBRTC_VIDEO_RECOVERY_AFTER_ERROR_PER_MIN,
- err_per_min);
float percent = (mTotalLossTime.ToSeconds()*100)/callDelta.ToSeconds();
CSFLogError(logTag, "Video error time percentage %f%%", percent);
- Telemetry::Accumulate(Telemetry::WEBRTC_VIDEO_DECODE_ERROR_TIME_PERMILLE,
- static_cast<uint32_t>(percent*10));
}
}
#endif
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