12 Commits

Author SHA1 Message Date
Yves Gerey
59cfd35438 Address vptr race condition while PeerConnection is destructed.
Auxiliary threads (worker, network) are still active
while PeerConnection is destructed, leading to race condition
in tests such as:
  * RTCStatsIntegrationTest.GetStatsFromCaller
  * RTCStatsIntegrationTest.GetsStatsWhileDestroyingPeerConnection

This CL prevents the conflict to happen by explicitly
closing the PeerConnection.

Bug: webrtc:9847
Change-Id: I40880bb9b193201711031b8c4563c6bbd4983c71
Reviewed-on: https://webrtc-review.googlesource.com/c/104820
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25801}
2018-11-27 14:06:37 +00:00
Niels Möller
f06f923ef0 Delete almost all use of MediaConstraintsInterface in the PeerConnection API
Bug: webrtc:9239
Change-Id: I04f4370f624346bf72c7e4e090b57987b558213b
Reviewed-on: https://webrtc-review.googlesource.com/74420
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24396}
2018-08-23 07:14:37 +00:00
Artem Titov
e41c433502 Move sigslot to proper third_party directory
Extract sigslot into separate target and move it to proper third_party
directory.

Bug: webrtc:8366
Change-Id: Id2e0712bd020bfad811947803c94553dce06d976
Reviewed-on: https://webrtc-review.googlesource.com/84141
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24099}
2018-07-25 14:53:33 +00:00
Yves Gerey
665174fdbb Reformat the WebRTC code base
Running clang-format with chromium's style guide.

The goal is n-fold:
 * providing consistency and readability (that's what code guidelines are for)
 * preventing noise with presubmit checks and git cl format
 * building on the previous point: making it easier to automatically fix format issues
 * you name it

Please consider using git-hyper-blame to ignore this commit.

Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
2018-06-19 14:00:39 +00:00
Niels Möller
2d02e085de Delete deprecated CreateAudioSource method, with constraints.
Bug: webrtc:9239
Change-Id: I5025b7fd103247e0426ceabedc1216a4f0f0ab34
Reviewed-on: https://webrtc-review.googlesource.com/76560
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23501}
2018-06-04 08:19:30 +00:00
Harald Alvestrand
5081c0cc6d Change error handlers for Set*Description to use RTCError
Needed in order to return error codes to Chromium.

Bug: chromium:819629, chromium:589455
Change-Id: Iab22250db62a348eee21c6d8bfc44020a7380586
Reviewed-on: https://webrtc-review.googlesource.com/60522
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22367}
2018-03-09 15:37:34 +00:00
Steve Anton
fc8537143f Crash if PeerConnection methods are called with the wrong SdpSemantics.
Bug: None
Change-Id: I111098215ec83fdf97f9a5232efef6a4af329ddf
Reviewed-on: https://webrtc-review.googlesource.com/59262
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22262}
2018-03-02 03:12:49 +00:00
Steve Anton
191c39f307 Parameterize PeerConnection end to end tests for Unified Plan
Bug: webrtc:8765
Change-Id: If4b797be7876a7680e99c698631c29b412f7a455
Reviewed-on: https://webrtc-review.googlesource.com/41540
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21755}
2018-01-25 04:03:18 +00:00
Steve Anton
a3a92c264b Replace string type with SdpType enum
This moves all WebRTC internal code from using
SessionDescriptionInterface::type() which returns a string and
from using CreateSessionDescription with a string type parameter.

Bug: webrtc:8613
Change-Id: I1cdd93dc4b26dec157e22476fdac569d5da2810a
Reviewed-on: https://webrtc-review.googlesource.com/29500
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21147}
2017-12-07 19:08:47 +00:00
Steve Anton
36b29d1df3 Enable cpplint in pc/
Enable cpplint check in the PRESUBMIT for pc/ and fix all existing
warnings.

Bug: webrtc:5583
Change-Id: If39994692ab6f6f3c83c74f23850f02fdfe810e8
Reviewed-on: https://webrtc-review.googlesource.com/16540
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20482}
2017-10-30 18:08:29 +00:00
Mirko Bonadei
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00
Mirko Bonadei
bb547203bf Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00