Commit Graph

9 Commits

Author SHA1 Message Date
c20978e581 Rename webrtc/base -> webrtc/rtc_base
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
NOTRY=True
NOTREECHECKS=True
TBR=kwiberg@webrtc.org, kjellander@webrtc.org

Bug: webrtc:7634
Change-Id: I3cca0fbaa807b563c95979cccd6d1bec32055f36
Reviewed-on: https://chromium-review.googlesource.com/562156
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18919}
2017-07-06 19:11:40 +00:00
a80c16a67c Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
This reverts commit c3771cc4d37f5573fe53b7c7cff295a4f0f9560f.
(breaks downstream internal project)

BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2972463002 .
Cr-Commit-Position: refs/heads/master@{#18873}
2017-07-01 14:48:18 +00:00
c3771cc4d3 Update includes for webrtc/{base => rtc_base} rename (2/3)
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}
2017-06-30 20:42:44 +00:00
f79ade1320 Revert "Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ )"
This reverts commit d72098a41971833e210bfdcffaab7a18ced4775f.

BUG=webrtc:5079

Review-Url: https://codereview.webrtc.org/2915263002
Cr-Commit-Position: refs/heads/master@{#18411}
2017-06-02 13:44:03 +00:00
d72098a419 Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ )
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: e80f4c91d0

TBR=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}
2017-06-01 15:54:47 +00:00
e80f4c91d0 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}
2017-06-01 14:29:30 +00:00
804c1af48b Move trackmediainfomap files from api/ to pc/.
It looks like this was left out of the original api/pc move CL since it
had been added recently.

BUG=webrtc:5883
TBR=ossu@webrtc.org

Review-Url: https://codereview.webrtc.org/2690793003
Cr-Commit-Position: refs/heads/master@{#16560}
2017-02-12 03:07:31 +00:00
338f78ac95 RTCIceCandidatePairStats.available[Outgoing/Incoming]Bitrate collected.
Collected for current pairs, undefined for other pairs. This is the
same as the old stats' VideoBwe.googAvailable[Send/Receive]Bandwidth.

NOTE: The value this is based on for incoming bitrate is not set. This
CL wires it up but has a TODO that the incoming bitrate needs to be
collected properly. (Same problem for both old and new stats.)

Spec: https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-availableoutgoingbitrate
Discussion: https://github.com/w3c/webrtc-stats/issues/112#issuecomment-277167781

BUG=webrtc:7062

Review-Url: https://codereview.webrtc.org/2675923002
Cr-Commit-Position: refs/heads/master@{#16472}
2017-02-07 14:41:21 +00:00
7bb87ee4e8 Create //webrtc/api:libjingle_peerconnection_api + refactorings.
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}
2017-01-23 12:56:25 +00:00