streams.
When addTrack/removeTrack is used instead of addStream/removeStream
we an end up with tracks that are not contained within any local or
remote stream.
If all track IDs are not mapped when we produce RTCRTPStreamStats
we'll hit a DCHECK.
BUG=chromium:741638
Review-Url: https://codereview.webrtc.org/2978793002
Cr-Commit-Position: refs/heads/master@{#18991}
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`
BUG=webrtc:7634
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
Review-Url: https://codereview.webrtc.org/2969623003
Cr-Commit-Position: refs/heads/master@{#18870}
Reason for revert:
Broken downstream projects
Original issue's description:
> Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls.
>
> Prior to this CL Call::Stats were collected via WebRtcVideoEngine2, but not via WebRtcVoiceEngine, causing these stats to be missing for audio-only calls. Call lives on the peerconnection/session level and should only be collected once independent on how many streams we have.
>
> BUG=webrtc:5079
> R=deadbeef@webrtc.org, hbos@webrtc.org
>
> Review-Url: https://codereview.webrtc.org/2863123002 .
> Cr-Commit-Position: refs/heads/master@{#18384}
> Committed: e80f4c91d0TBR=hbos@webrtc.org,deadbeef@webrtc.org,holmer@google.com,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5079
Review-Url: https://codereview.webrtc.org/2916793003
Cr-Commit-Position: refs/heads/master@{#18386}
Prior to this CL Call::Stats were collected via WebRtcVideoEngine2, but not via WebRtcVoiceEngine, causing these stats to be missing for audio-only calls. Call lives on the peerconnection/session level and should only be collected once independent on how many streams we have.
BUG=webrtc:5079
R=deadbeef@webrtc.org, hbos@webrtc.org
Review-Url: https://codereview.webrtc.org/2863123002 .
Cr-Commit-Position: refs/heads/master@{#18384}
This is a problem if a data channel is re-opened or a new data channel
occupies the same space in memory as a previously closed data channel.
Unittest updated to cover this (failed before fix, now passes).
BUG=webrtc:7181
Review-Url: https://codereview.webrtc.org/2746393003
Cr-Commit-Position: refs/heads/master@{#17304}
The value is being moved:
https://github.com/w3c/webrtc-stats/pull/167
Stop collecting this value. Our previous value was incorrect, our RTT
value was a smoothed value based on STUN pings but the spec says it
should be based on RTCP timestamps in RTCP Receiver Report (RR) on
inbound streams with isRemote=true (not supported).
Updated some bug references.
BUG=webrtc:7065, webrtc:7066
Review-Url: https://codereview.webrtc.org/2722633005
Cr-Commit-Position: refs/heads/master@{#16931}
Collected in accordance with the spec:
https://w3c.github.io/webrtc-stats/#candidatepair-dict*
totalRoundTripTime is collected as the sum of rtt measurements, it was
previously not collected.
currentRoundTripTime is collected as the latest rtt measurement, it
was previously collected as a smoothed value, which was incorrect.
Connection is updated to collect these values which are surfaced
through ConnectionInfo.
BUG=webrtc:7062, webrtc:7204
Review-Url: https://codereview.webrtc.org/2719523002
Cr-Commit-Position: refs/heads/master@{#16905}
Connection::nominated() is updated to mean
(remote_nomination_ || acked_nomination_), which means both a
controlling and controlled agent can be said to be "nominated".
Previously this was (remote_nomination_ > 0) which only applies to the
controlling agent.
PortTest.TestNomination added to test nomination values and nomination
stat.
This value is surfaced through cricket::ConnectionInfo::nominated.
RTCStatsCollector uses this value in its collection of
RTCIceCandidatePairStats.
RTCStatsCollectorTest.CollectRTCIceCandidatePairStats updated to test
that ConnectionInfo::nominated is surfaced using mocks.
rtcstats_integrationtest.cc updated to expect nomination set without
using mocks.
Spec: https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-nominated
BUG=webrtc:7062, webrtc:7204
Review-Url: https://codereview.webrtc.org/2709293004
Cr-Commit-Position: refs/heads/master@{#16855}
Reason for revert:
Breaks downstream build.
Original issue's description:
> RTCInboundRTPStreamStats.qpSum collected.
>
> This was previously only collected for local tracks
> (RTCOutboundRTPStreamStats.qpSum).
>
> Spec: https://w3c.github.io/webrtc-stats/#dom-rtcrtpstreamstats-qpsum
>
> This CL also improves some testing in rtcstatscollector_unittest.cc.
> Default and non-default values are tested in the same unittests,
> removing the test that was specific to default-values, which was
> otherwise code duplication.
>
> BUG=webrtc:7065
>
> Review-Url: https://codereview.webrtc.org/2675943002
> Cr-Commit-Position: refs/heads/master@{#16477}
> Committed: cd195bea5eTBR=sakal@webrtc.org,hta@webrtc.org,hbos@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7065
Review-Url: https://codereview.webrtc.org/2687483002 .
Cr-Commit-Position: refs/heads/master@{#16479}
This was previously only collected for local tracks
(RTCOutboundRTPStreamStats.qpSum).
Spec: https://w3c.github.io/webrtc-stats/#dom-rtcrtpstreamstats-qpsum
This CL also improves some testing in rtcstatscollector_unittest.cc.
Default and non-default values are tested in the same unittests,
removing the test that was specific to default-values, which was
otherwise code duplication.
BUG=webrtc:7065
Review-Url: https://codereview.webrtc.org/2675943002
Cr-Commit-Position: refs/heads/master@{#16477}
Refactor how |codec_id| is set, remove outdated TODO, update comments
with new bugs IDs.
BUG=webrtc:7061
Review-Url: https://codereview.webrtc.org/2670343002
Cr-Commit-Position: refs/heads/master@{#16467}
These structs will be used for ORTC objects (and their WebRTC
equivalents).
This CL also introduces some minor changes to the existing implemented
structs:
- max_bitrate_bps uses rtc::Optional instead of "-1 means unset"
- "mime_type" turned into "name"/"kind" (which can be used to form the
MIME type string, if needed).
- clock_rate and channels changed to rtc::Optional, since they will
need to be for RtpSender.send().
- Renamed "channels" to "num_channels" (the ORTC name, which I prefer).
BUG=webrtc:7013, webrtc:7112
Review-Url: https://codereview.webrtc.org/2651883010
Cr-Commit-Position: refs/heads/master@{#16437}
Create a new target //webrtc/api:libjingle_peerconnection_api and start moving
things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection
to //webrtc/pc:libjingle_peerconnection.
Moved the RTCStatsCollectorCallback into its own header file, so that
PeerConnectionInterface can include that instead of pulling in
RTCStatsCollector and PeerConnection and everything.
Separated cricket::MediaType into its own header/source set, so that it
can be used in the api.
BUG=webrtc:5883
Review-Url: https://codereview.webrtc.org/2514883002
Cr-Commit-Position: refs/heads/master@{#16210}
