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Issue 657855 link

Starred by 3 users

Issue metadata

Status: Duplicate
Merged: issue webrtc:7065
Owner:
Closed: Jan 2017
Cc:
Components:
EstimatedDays: ----
NextAction: ----
OS: ----
Pri: 3
Type: Bug

Blocked on:
issue 657854
issue webrtc:6758
issue webrtc:6826

Blocking:
issue 627816



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RTCPeerConnection.getStats: RTCInboundRTPStreamStats

Project Member Reported by hbos@chromium.org, Oct 20 2016

Issue description

Add RTCInboundRTPStreamStats and all of its members.
https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*

dictionary RTCInboundRTPStreamStats : RTCRTPStreamStats {
    unsigned long      packetsReceived;
    unsigned long long bytesReceived;
    unsigned long      packetsLost;
    double             jitter;
    double             fractionLost;
    unsigned long      packetsDiscarded;
    unsigned long      packetsRepaired;
    unsigned long      burstPacketsLost;
    unsigned long      burstPacketsDiscarded;
    unsigned long      burstLossCount;
    unsigned long      burstDiscardCount;
    double             burstLossRate;
    double             burstDiscardRate;
    double             gapLossRate;
    double             gapDiscardRate;
    unsigned long      framesDecoded;  // Since 14 December 2016 Draft
};
 
Project Member

Comment 2 by bugdroid1@chromium.org, Nov 1 2016

The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/4ed075034a676aad8ec9d65444cb914d4a20b3af

commit 4ed075034a676aad8ec9d65444cb914d4a20b3af
Author: perkj <perkj@webrtc.org>
Date: Tue Nov 01 06:50:53 2016

Revert of RTCInboundRTPStreamStats added. (patchset #4 id:100001 of https://codereview.webrtc.org/2452043002/ )

Reason for revert:
Dependend cl Breaks Chrome FYI.
peerconnection_unittest anropar RTCStatsReport::Create without  parameters.

Original issue's description:
> RTCInboundRTPStreamStats[1] added.
>
> Not all stats are collected in this CL, this must be addressed before
> closing the issue.
>
> [1] https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*
>
> BUG= chromium:627816 ,  chromium:657855 ,  chromium:657854 
>
> Committed: https://crrev.com/0d7bf169402ea9345d163998f4f7df89229ac470
> Cr-Commit-Position: refs/heads/master@{#14861}

TBR=hta@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG= chromium:627816 ,  chromium:657855 ,  chromium:657854 

Review-Url: https://codereview.webrtc.org/2470683002
Cr-Commit-Position: refs/heads/master@{#14862}

[modify] https://crrev.com/4ed075034a676aad8ec9d65444cb914d4a20b3af/webrtc/api/rtcstatscollector.cc
[modify] https://crrev.com/4ed075034a676aad8ec9d65444cb914d4a20b3af/webrtc/api/rtcstatscollector.h
[modify] https://crrev.com/4ed075034a676aad8ec9d65444cb914d4a20b3af/webrtc/api/rtcstatscollector_unittest.cc
[modify] https://crrev.com/4ed075034a676aad8ec9d65444cb914d4a20b3af/webrtc/api/stats/rtcstats_objects.h
[modify] https://crrev.com/4ed075034a676aad8ec9d65444cb914d4a20b3af/webrtc/stats/rtcstats_objects.cc

Project Member

Comment 4 by bugdroid1@chromium.org, Nov 22 2016

The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/820f578f2bb3585cc48876fa9e594a6a6b0dd174

commit 820f578f2bb3585cc48876fa9e594a6a6b0dd174
Author: hbos <hbos@webrtc.org>
Date: Tue Nov 22 11:16:50 2016

RTCInboundRTPStreamStats's [fir/pli/nack]_count are collected for video.

Previously this was only collected for RTCOutboundRTPStreamStats video,
with no comment saying it was missing for Inbound. (nack_count should be
collected vor audio as well but this is currently not available - there
is already an existing comment about this in rtcstats_objects.h.)

BUG= chromium:657855 ,  chromium:657854 ,  chromium:627816 

Review-Url: https://codereview.webrtc.org/2515293002
Cr-Commit-Position: refs/heads/master@{#15185}

[modify] https://crrev.com/820f578f2bb3585cc48876fa9e594a6a6b0dd174/webrtc/api/rtcstatscollector.cc
[modify] https://crrev.com/820f578f2bb3585cc48876fa9e594a6a6b0dd174/webrtc/api/rtcstatscollector_unittest.cc

Comment 5 by hbos@webrtc.org, Nov 23 2016

Blockedon: webrtc:6758

Comment 6 by hbos@chromium.org, Nov 30 2016

Blockedon: 669877

Comment 7 by hbos@webrtc.org, Dec 8 2016

Blockedon: webrtc:6826

Comment 9 by hbos@chromium.org, Dec 12 2016

Blockedon: -669877

Comment 10 by hbos@chromium.org, Dec 21 2016

Description: Show this description
Project Member

Comment 11 by bugdroid1@chromium.org, Jan 2 2017

The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/6769c4941817ac3d5b967d0aafa1304c06d0672e

commit 6769c4941817ac3d5b967d0aafa1304c06d0672e
Author: hbos <hbos@webrtc.org>
Date: Mon Jan 02 16:35:13 2017

RTC[In/Out]boundRTPStreamStats: qpSum,framesDecoded,framesEncoded added.

Recently added to the spec:
RTCRTPStreamStats.qpSum - https://w3c.github.io/webrtc-stats/#dom-rtcrtpstreamstats-qpsum
RTCInboundRTPStreamStats.framesDecoded - https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-framesdecoded
RTCOutboundRTPStreamStats.framesEncoded - https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-framesencoded

These are added and collected. However, the qpSum is only collected in
the outbound case. It should be collected in the inbound case before
closing  crbug.com/657855 

BUG= chromium:657854 ,  chromium:657855 ,  chromium:657856 

Review-Url: https://codereview.webrtc.org/2588373005
Cr-Commit-Position: refs/heads/master@{#15872}

[modify] https://crrev.com/6769c4941817ac3d5b967d0aafa1304c06d0672e/webrtc/api/rtcstats_integrationtest.cc
[modify] https://crrev.com/6769c4941817ac3d5b967d0aafa1304c06d0672e/webrtc/api/rtcstatscollector.cc
[modify] https://crrev.com/6769c4941817ac3d5b967d0aafa1304c06d0672e/webrtc/api/rtcstatscollector_unittest.cc
[modify] https://crrev.com/6769c4941817ac3d5b967d0aafa1304c06d0672e/webrtc/api/stats/rtcstats_objects.h
[modify] https://crrev.com/6769c4941817ac3d5b967d0aafa1304c06d0672e/webrtc/stats/rtcstats_objects.cc

Project Member

Comment 12 by bugdroid1@chromium.org, Jan 2 2017

The following revision refers to this bug:
  https://chromium.googlesource.com/chromium/src.git/+/a3a849d6f9efef1f90688ef3d11f8ca75618ac3a

commit a3a849d6f9efef1f90688ef3d11f8ca75618ac3a
Author: hbos <hbos@chromium.org>
Date: Mon Jan 02 16:43:12 2017

RTCPeerConnection.getStats: Whitelist updated after spec changed.

This is the whitelist used in the Chromium integration test.

New members:
- RTCRTPStreamStats.qpSum: https://w3c.github.io/webrtc-stats/#dom-rtcrtpstreamstats-qpsum
- RTCInboundRTPStreamStats.framesDecoded: https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-framesdecoded
- RTCOutboundRTPStreamStats.framesEncoded: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-framesencoded
- RTCIceCandidateStats.transportId: https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatestats-transportid
- RTCIceCandidateStats.isRemote: https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatestats-isremote
- RTCIceCandidateStats.deleted: https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatestats-deleted

Renamed members, removed old names (https://codereview.webrtc.org/2576383002/):
- RTCIceCandidatePairStats.totalRtt replaced by totalRoundTripTime: https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-totalroundtriptime
- RTCIceCandidatePairStats.currentRtt replaced by currentRoundTripTime: https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-currentroundtriptime

NOTRY=True
BUG= chromium:627816 ,  chromium:657854 ,  chromium:657855 ,  chromium:657856 ,  chromium:632723 

Review-Url: https://codereview.chromium.org/2590293004
Cr-Commit-Position: refs/heads/master@{#441073}

[modify] https://crrev.com/a3a849d6f9efef1f90688ef3d11f8ca75618ac3a/chrome/test/data/webrtc/peerconnection_getstats.js

Comment 13 by hbos@chromium.org, Jan 11 2017

Issue raised in https://codereview.webrtc.org/2610843003/#msg32 - https://codereview.webrtc.org/2610843003/#msg33:

Observation: When setting up a Hangouts call and invoking the new getStats, there are temporarily RTCInboundRTPStreamStats that don't have RTCMediaStreamTrackStats.

deadbeef: "The issue you saw in testing may be due to SSRCs being unsignaled, and no packets received yet? In which case it may make sense to have no RTCInboundRTPStreamStats yet."
Project Member

Comment 14 by bugdroid1@chromium.org, Jan 16 2017

The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/84abeb1d37ee27147674cd94a5606bd4ac725d9a

commit 84abeb1d37ee27147674cd94a5606bd4ac725d9a
Author: hbos <hbos@webrtc.org>
Date: Mon Jan 16 14:16:44 2017

RTC[In/Out]boundRTPStreamStats.mediaTrackId collected.

Based on the mapping between [Audio/Video]TrackInterface and
[Voice/Video][Sender/Receiver]Info.

The IDs of RTCMediaStreamTrackStats are updated to distinguish between
local and remote cases. Previously, if local and remote cases had the
same label only one of them would be included in the report (bug).

BUG= webrtc:6758 ,  chromium:657854 ,  chromium:657855 ,  chromium:657856 ,  chromium:627816 

Review-Url: https://codereview.webrtc.org/2610843003
Cr-Commit-Position: refs/heads/master@{#16095}

[modify] https://crrev.com/84abeb1d37ee27147674cd94a5606bd4ac725d9a/webrtc/api/rtcstats_integrationtest.cc
[modify] https://crrev.com/84abeb1d37ee27147674cd94a5606bd4ac725d9a/webrtc/api/rtcstatscollector.cc
[modify] https://crrev.com/84abeb1d37ee27147674cd94a5606bd4ac725d9a/webrtc/api/rtcstatscollector.h
[modify] https://crrev.com/84abeb1d37ee27147674cd94a5606bd4ac725d9a/webrtc/api/rtcstatscollector_unittest.cc
[modify] https://crrev.com/84abeb1d37ee27147674cd94a5606bd4ac725d9a/webrtc/api/stats/rtcstats_objects.h
[modify] https://crrev.com/84abeb1d37ee27147674cd94a5606bd4ac725d9a/webrtc/api/statscollector_unittest.cc
[modify] https://crrev.com/84abeb1d37ee27147674cd94a5606bd4ac725d9a/webrtc/api/test/mock_peerconnection.h

Comment 15 by hbos@chromium.org, Jan 26 2017

Mergedinto: webrtc:7065
Status: Duplicate (was: Started)
This stats dictionary is partially supported (see landed CLs in this thread).
To better track what's left, a webrtc bug has been created to track the remaining work related to this dictionary - merging into that one.

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