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43cb716e55eab1cf1ae8afd2ac79b51a604d0fa5
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5 Commits

Author SHA1 Message Date
asapersson
43cb716e55 Add ToString method to AggregatedStats and log stats at the end of a call. 
BUG=webrtc:5283

Review-Url: https://codereview.webrtc.org/2494423002
Cr-Commit-Position: refs/heads/master@{#15088}
 2016-11-15 16:20:54 +00:00
asapersson
1d02d3e5e6 Remove RTC_LOGGED_* macro. 
BUG=

Review-Url: https://codereview.webrtc.org/2326843003
Cr-Commit-Position: refs/heads/master@{#14174}
 2016-09-10 05:40:34 +00:00
asapersson
ce2e13602e Update AvgCounter to have the ability to include last period metric for subsequent intervals without samples (e.g. for non-periodic updated stats). 
Integrate AvgCounter to be used for BWE stats in call.

Fixes for stats regression in:
WebRTC.Call.EstimatedSendBitrateInKbps
WebRTC.Call.PacerBitrateInKbps

Example:
BWE for a 15 seconds long call (with intervals of 1 sec):
|300|400|500|600|600|600|600| 0 | 0 | 0 | 0 | 0 |800|800|800|  // 0 - network state down

Reported via OnNetworkChanged:
|300|400|500|600| x | x | x | 0 | x | x | x | x |800| x | x |  // x - empty interval, 0 -> pauses stats

Stats:
|300|400|500|600|600|600|600| - | - | - | - | - |800|800|800|  // x -> last value used (intervals during pause ignored)

AvgCounter uses the average of samples within an interval (interval length is 2 sec).

BUG=webrtc:6244

Review-Url: https://codereview.webrtc.org/2307913002
Cr-Commit-Position: refs/heads/master@{#14147}
 2016-09-09 07:13:39 +00:00
asapersson
40f54003d6 Start integrating StatsCounter class. 
SampleCounter replaced with AvgCounter for metric WebRTC.Video.SendDelayInMs (removes duplicated code of SampleCounter).

BUG=webrtc:5283

Review-Url: https://codereview.webrtc.org/2013403002
Cr-Commit-Position: refs/heads/master@{#13081}
 2016-06-09 07:09:27 +00:00
asapersson
35151f35ec Add histogram stats for average send delay of sent packets for a sent video stream. The delay is measured from a packet is sent to the transport until leaving the socket. 
- "WebRTC.Video.SendDelayInMs"

Change so that PacketOption packet id is always set in RtpSender (if having a TransportSequenceNumberAllocator).
Add SendDelayStats class for computing delays.
Add SendPacketObserver to RtpRtcp config and register SendDelayStats as observer.
Wire up OnSentPacket to SendDelayStats.

BUG=webrtc:5215

Review-Url: https://codereview.webrtc.org/1478253002
Cr-Commit-Position: refs/heads/master@{#12600}
 2016-05-03 06:44:11 +00:00