16 Commits

Author SHA1 Message Date
mbonadei
539d104e3e Revert of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt (patchset #2 id:20001 of https://codereview.webrtc.org/2964863002/ )
Reason for revert:
It breaks a downstream project.

Original issue's description:
> Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on the worker thread.
>
> Added documentation of thread expectations for video tracks and sources to the API.
>
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2964863002
> Cr-Commit-Position: refs/heads/master@{#18938}
> Committed: f1377f7222

TBR=deadbeef@webrtc.org,noahric@chromium.org,yujo@chromium.org,perkj@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=None

Review-Url: https://codereview.webrtc.org/2979493003
Cr-Commit-Position: refs/heads/master@{#18942}
2017-07-10 09:40:49 +00:00
perkj
f1377f7222 Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on the worker thread.
Added documentation of thread expectations for video tracks and sources to the API.

BUG=None

Review-Url: https://codereview.webrtc.org/2964863002
Cr-Commit-Position: refs/heads/master@{#18938}
2017-07-07 23:38:28 +00:00
Edward Lemur
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
Henrik Kjellander
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
kjellander
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
Alex Narest
42308f615c Fix uploading of available send bitrate statistics.
BUG=webrtc:5079
R=stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/2943073002 .
Cr-Commit-Position: refs/heads/master@{#18664}
2017-06-19 15:58:15 +00:00
stefan
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
charujain
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
Stefan Holmer
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
nisse
eaabdf6259 Delete MediaController class, move Call ownership to PeerConnection.
BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2794943002
Cr-Commit-Position: refs/heads/master@{#18026}
2017-05-05 09:23:02 +00:00
deadbeef
112b2e99d8 Switching some interfaces to use std::unique_ptr<>.
This helps show where ownership is transfered between objects.

Specifically, this CL wraps cricket::VideoCapturer, MediaEngineInterface
and DataEngineInterface in unique_ptr.

BUG=None
TBR=magjed@webrtc.org

Review-Url: https://codereview.webrtc.org/2685093002
Cr-Commit-Position: refs/heads/master@{#16548}
2017-02-11 04:13:37 +00:00
sakal
cc452e1179 Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
Reason for revert:
Fix the problem.

Original issue's description:
> Revert of Add QP sum stats for received streams. (patchset #10 id:180001 of https://codereview.webrtc.org/2649133005/ )
>
> Reason for revert:
> Breaks downstream build.
>
> Original issue's description:
> > Add QP sum stats for received streams.
> >
> > This is not implemented yet in any of the decoders.
> >
> > BUG=webrtc:6541
> >
> > Review-Url: https://codereview.webrtc.org/2649133005
> > Cr-Commit-Position: refs/heads/master@{#16475}
> > Committed: ff0e72fd16
>
> TBR=hta@webrtc.org,hbos@webrtc.org,sprang@webrtc.org,magjed@webrtc.org,stefan@webrtc.org,sakal@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6541
>
> Review-Url: https://codereview.webrtc.org/2680893002 .
> Cr-Commit-Position: refs/heads/master@{#16480}
> Committed: 69fb2cca4d

TBR=hta@webrtc.org,hbos@webrtc.org,sprang@webrtc.org,magjed@webrtc.org,stefan@webrtc.org,skvlad@webrtc.org
BUG=webrtc:6541

Review-Url: https://codereview.webrtc.org/2681663005
Cr-Commit-Position: refs/heads/master@{#16511}
2017-02-09 12:53:45 +00:00
skvlad
69fb2cca4d Revert of Add QP sum stats for received streams. (patchset #10 id:180001 of https://codereview.webrtc.org/2649133005/ )
Reason for revert:
Breaks downstream build.

Original issue's description:
> Add QP sum stats for received streams.
>
> This is not implemented yet in any of the decoders.
>
> BUG=webrtc:6541
>
> Review-Url: https://codereview.webrtc.org/2649133005
> Cr-Commit-Position: refs/heads/master@{#16475}
> Committed: ff0e72fd16

TBR=hta@webrtc.org,hbos@webrtc.org,sprang@webrtc.org,magjed@webrtc.org,stefan@webrtc.org,sakal@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6541

Review-Url: https://codereview.webrtc.org/2680893002 .
Cr-Commit-Position: refs/heads/master@{#16480}
2017-02-07 18:59:25 +00:00
sakal
ff0e72fd16 Add QP sum stats for received streams.
This is not implemented yet in any of the decoders.

BUG=webrtc:6541

Review-Url: https://codereview.webrtc.org/2649133005
Cr-Commit-Position: refs/heads/master@{#16475}
2017-02-07 15:15:17 +00:00
deadbeef
f534659ee6 Adding ability for BaseChannel to use PacketTransportInterface.
... As opposed to DtlsTransportInternal.

The code is suboptimal right now, storing two pointers to the different
interfaces. This will all be cleaned up when we have an "RtpTransport"
abstraction that BaseChannel can use.

This CL also cleans up the "fake transport" classes a bit, and gives
them their own header files.

BUG=None

Review-Url: https://codereview.webrtc.org/2648233003
Cr-Commit-Position: refs/heads/master@{#16258}
2017-01-25 05:51:21 +00:00
ossu
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