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author | Moonchild <mcwerewolf@gmail.com> | 2018-09-03 18:26:19 +0200 |
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committer | GitHub <noreply@github.com> | 2018-09-03 18:26:19 +0200 |
commit | 2e00eb87ef299e6eb7521670e6a6720fee19f5fc (patch) | |
tree | c44670a25d942a672951e430499f70978ec7d337 /media/webrtc/signaling/src/media-conduit/AudioConduit.cpp | |
parent | 45f9a0daad81d1c6a1188b3473e5f0c67d27c0aa (diff) | |
parent | ab961aeb54335fd07c66de2e3b8c3b6af6f89ea2 (diff) | |
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Merge pull request #745 from MoonchildProductions/Kill-Telemetry
Remove all C++ Telemetry Accumulation calls.
Diffstat (limited to 'media/webrtc/signaling/src/media-conduit/AudioConduit.cpp')
-rwxr-xr-x | media/webrtc/signaling/src/media-conduit/AudioConduit.cpp | 9 |
1 files changed, 0 insertions, 9 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); |