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authorwolfbeast <mcwerewolf@gmail.com>2018-09-03 10:11:38 +0200
committerwolfbeast <mcwerewolf@gmail.com>2018-09-03 10:11:38 +0200
commitab961aeb54335fd07c66de2e3b8c3b6af6f89ea2 (patch)
treec44670a25d942a672951e430499f70978ec7d337 /media/webrtc/signaling/src/media-conduit
parent45f9a0daad81d1c6a1188b3473e5f0c67d27c0aa (diff)
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Remove all C++ Telemetry Accumulation calls.
This creates a number of stubs and leaves some surrounding code that may be irrelevant (eg. recorded time stamps, status variables). Stub resolution/removal should be a follow-up to this.
Diffstat (limited to 'media/webrtc/signaling/src/media-conduit')
-rwxr-xr-xmedia/webrtc/signaling/src/media-conduit/AudioConduit.cpp9
-rw-r--r--media/webrtc/signaling/src/media-conduit/CodecStatistics.cpp8
2 files changed, 0 insertions, 17 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