Comments (6)
I vote for (and have assumed) ms
from webrtc-stats.
Since the type is double, I feel the milliseconds for the integer part may be the right resolution.
That seems counter-intuitive to me. Had you said "Since the type is integer", I would have bought ms.
But with floating point, why not just assume the base unit?
FWIW we've implemented jitter
and mozRtt
since Firefox 30 (both in seconds).
from webrtc-stats.
currently the spec is not clear on what the unit for all RTT measurements are. In one place it says (seconds), and for the other no units are presented.
@vr000m The place where it says "seconds" was committed by you on Aug 29, 2014.
The place with no mention of unit was committed by you on Feb 2, 2015.
So for at least 6 months it was consistent. ;-)
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I'll revise -- with floating point, seconds makes sense.
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@jan-ivar, my implementation of STUN-PATH-CHAR draft uses milliseconds, although the spec itself does not say what it should be measured in.
If we have converged on seconds, I'll fix my PR to seconds and move ahead.
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PR #44 (wrong title) will set it to seconds.
from webrtc-stats.
Related Issues (20)
- Conformance criteria incompatible with standards-track HOT 7
- RTX and FEC stats are incomplete. HOT 12
- Should we flatten the hierarchy? HOT 5
- Change playoutdelay to jitterbuffertarget HOT 2
- Add clarification that inbound-rtp.bytesReceived includes retransmissions. HOT 6
- RTCAudioSourceStats.audioLevel - what algorithm described in totalAudioEnergy? HOT 3
- Delete the obsolete section, "track" and "stream" stats - they don't exist anymore HOT 3
- Improve test coverage HOT 1
- add framesInput stats to rtcoutboundrtpstreamstats HOT 6
- Stats example uses confusing variable names
- Exposing audio interruption metrics to JavaScript
- Define a mechanism for setting the trigger duration for a video freeze
- Sender-side packetsReceived can be negative from garbage RRs HOT 1
- Change "Does not" to "must not"
- Make "not present" reference "map/exist"
- Is codec.sdpFmtpLine present when there is no fmtp line in the SDP? HOT 2
- Lifetime of outbound-rtp should start BEFORE first packet is sent HOT 3
- Should RTCRtptransceiver.stop() cause inbound-rtp stats to disappear?
- RTCCodecStats.clockRate - media sampling rate or the codec clock rate?
- RTCStats.timestamp - fingerprinting and since epoch
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