| f475d365a25036725c3f545f57de59d2cc902d17 |
|
09-Jan-2016 |
Taylor Brandstetter <deadbeef@webrtc.org> |
Properly handle different transports having different SSL roles.
This meant splitting "transport_options" into audio/video/data options,
for when creating the answer, and giving "GetSslRole" a "transport_name"
parameter so we can retrieve the current role on a per-transport basis.
BUG=webrtc:4525
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1516993002 .
Cr-Commit-Position: refs/heads/master@{#11192}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 60ca31bf5d206ff01b5441639806f7303365e162 |
|
04-Jan-2016 |
kjellander <kjellander@webrtc.org> |
Roll chromium_revision d66326c..4df108a (367167:367307)
The changes in https://chromium.googlesource.com/chromium/src/+/d66326c..4df108a/build/common.gypi
enables a lot more warnings, which have been disabled/fixed in this CL.
See tracking bugs for remaining work.
Change log: https://chromium.googlesource.com/chromium/src/+log/d66326c..4df108a
Full diff: https://chromium.googlesource.com/chromium/src/+/d66326c..4df108a
Changed dependencies:
* src/buildtools: https://chromium.googlesource.com/chromium/buildtools.git/+log/fee7f1e..6d0c448
* src/third_party/libsrtp: https://chromium.googlesource.com/chromium/deps/libsrtp.git/+log/b8dd754..8a7662a
DEPS diff: https://chromium.googlesource.com/chromium/src/+/d66326c..4df108a/DEPS
No update to Clang.
BUG=webrtc:5397, webrtc:5398, webrtc:5399
TBR=hta@webrtc.org, perkj@webrtc.org
NOTRY=True
Review URL: https://codereview.webrtc.org/1553033002
Cr-Commit-Position: refs/heads/master@{#11147}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 0eb15ed7b806125774bd13fb214aeb403e2c6857 |
|
17-Dec-2015 |
kwiberg <kwiberg@webrtc.org> |
Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector
We can now use std::move instead!
This CL leaves the Pass methods in place; a follow-up CL will add deprecation annotations to them.
Review URL: https://codereview.webrtc.org/1460043002
Cr-Commit-Position: refs/heads/master@{#11064}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 04e9146e58bd68339b15ad651c9ee593d781e040 |
|
11-Dec-2015 |
Honghai Zhang <honghaiz@webrtc.org> |
Discard old-generation candidates when ICE restarts
The existing code only do so on the controlled side.
BUG=webrtc:5291
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1496693002 .
Cr-Commit-Position: refs/heads/master@{#10993}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 1d63dd0eaa44d13c5ae083200937b18bce2132ae |
|
02-Dec-2015 |
solenberg <solenberg@webrtc.org> |
- Remove cricket::VoiceChannel::PressDtmf(); AFAICT unused.
- Remove the DF_PLAY/DF_SEND flags, only allow sending.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1487393002
Cr-Commit-Position: refs/heads/master@{#10872}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 726b1f7a1467a33b1c3feedff84fca953f7f9c1d |
|
19-Nov-2015 |
perkj <perkj@webrtc.org> |
Removed dummy "mediastreamsignaling.h"
Review URL: https://codereview.webrtc.org/1460483005
Cr-Commit-Position: refs/heads/master@{#10717}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 653b8e02f22c9b6ba38be1cf4c0fa101894a9407 |
|
11-Nov-2015 |
deadbeef <deadbeef@webrtc.org> |
Reland of Adding the ability to change ICE servers through SetConfiguration. (patchset #1 id:1 of https://codereview.webrtc.org/1424803004/ )
Reason for revert:
Relanding with compile warning fixed.
Original issue's description:
> Revert of Adding the ability to change ICE servers through SetConfiguration. (patchset #7 id:120001 of https://codereview.webrtc.org/1391013007/ )
>
> Reason for revert:
> Caused compiler warning, breaking Chrome FYI bots.
>
> Original issue's description:
> > Adding the ability to change ICE servers through SetConfiguration.
> >
> > Added a SetIceServers method to PortAllocator. Also added a new
> > PeerConnection Initialize method that takes a PortAllocator, in the
> > hope that we can get rid of PortAllocatorFactoryInterface, since the
> > only substantial thing a factory does is convert the webrtc:: ICE
> > servers to cricket:: versions.
> >
> > Committed: https://crrev.com/d3b26d94399ff539db375a9b84010ee75479d4cf
> > Cr-Commit-Position: refs/heads/master@{#10420}
>
> TBR=pthatcher@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/18a944bf0ac9eed872dc009bd58e6bc12c946303
> Cr-Commit-Position: refs/heads/master@{#10421}
TBR=pthatcher@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1414313003
Cr-Commit-Position: refs/heads/master@{#10609}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| be57983f4bd875c39a229bab5112b32dad004057 |
|
10-Nov-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
Rename Maybe to Optional
And add examples of good and bad usage to the documentation.
R=aluebs@webrtc.org, henrik.lundin@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1432553007 .
Cr-Commit-Position: refs/heads/master@{#10588}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 102c6a61bc0b42dc0956d013530fc0213b7e881b |
|
30-Oct-2015 |
kwiberg <kwiberg@webrtc.org> |
Replace rtc::cricket::Settable with rtc::Maybe
The former is very similar to the latter, but less general (mostly in
naming).
This CL, which is the first to use Maybe at scale, also removes the implicit conversion from T to Maybe<T>, since it was agreed that the increased verbosity increased legibility.
Review URL: https://codereview.webrtc.org/1430433004
Cr-Commit-Position: refs/heads/master@{#10461}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 18a944bf0ac9eed872dc009bd58e6bc12c946303 |
|
27-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
Revert of Adding the ability to change ICE servers through SetConfiguration. (patchset #7 id:120001 of https://codereview.webrtc.org/1391013007/ )
Reason for revert:
Caused compiler warning, breaking Chrome FYI bots.
Original issue's description:
> Adding the ability to change ICE servers through SetConfiguration.
>
> Added a SetIceServers method to PortAllocator. Also added a new
> PeerConnection Initialize method that takes a PortAllocator, in the
> hope that we can get rid of PortAllocatorFactoryInterface, since the
> only substantial thing a factory does is convert the webrtc:: ICE
> servers to cricket:: versions.
>
> Committed: https://crrev.com/d3b26d94399ff539db375a9b84010ee75479d4cf
> Cr-Commit-Position: refs/heads/master@{#10420}
TBR=pthatcher@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1424803004
Cr-Commit-Position: refs/heads/master@{#10421}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| d3b26d94399ff539db375a9b84010ee75479d4cf |
|
27-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
Adding the ability to change ICE servers through SetConfiguration.
Added a SetIceServers method to PortAllocator. Also added a new
PeerConnection Initialize method that takes a PortAllocator, in the
hope that we can get rid of PortAllocatorFactoryInterface, since the
only substantial thing a factory does is convert the webrtc:: ICE
servers to cricket:: versions.
Review URL: https://codereview.webrtc.org/1391013007
Cr-Commit-Position: refs/heads/master@{#10420}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| c80741f8957b537e968397ac54ff5b5df8a2c318 |
|
22-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
Fixing some issues with the direction attribute of m-lines in offers.
By default, we'll now offer to receive if already receiving
(meaning that the last remote description contained a track).
Also, m-lines that are neither receiving nor sending are now correctly
marked "inactive".
Also moved some logic relating to default tracks out of webrtcsdp.cc,
such that now the direction seen by upper layers will always be
consistent with the consumed/produced SDP.
BUG=528089
Review URL: https://codereview.webrtc.org/1406803004
Cr-Commit-Position: refs/heads/master@{#10376}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| c1aeaf0dc37d96f31c92f893f4e30e7a5f7cc2b7 |
|
15-Oct-2015 |
stefan <stefan@webrtc.org> |
Wire up packet_id / send time callbacks to webrtc via libjingle.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1363573002
Cr-Commit-Position: refs/heads/master@{#10289}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| d59daf8023286d63a1b6c8af82eedb684181c1eb |
|
15-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
Merging BaseSession code into WebRtcSession.
After the TransportController CL, BaseSession does little more than
hold a state and an error, and act as an intermediary for the
TransportController. So it doesn't make sense for it to be its own
class.
Review URL: https://codereview.webrtc.org/1397973002
Cr-Commit-Position: refs/heads/master@{#10281}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| ab9b2d1516cad017c6e0236c468934582530c965 |
|
14-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
Reland of Moving MediaStreamSignaling logic into PeerConnection. (patchset #1 id:1 of https://codereview.webrtc.org/1403633005/ )
Reason for reland:
The original CL actually didn't break browser_tests; it was
just a coincidence that it started failing.
Original issue's description:
> Revert of Moving MediaStreamSignaling logic into PeerConnection. (patchset #10 id:180001 of https://codereview.webrtc.org/1393563002/ )
>
> Reason for revert:
> Broke browser_tests on Mac. Still need to investigate the cause.
>
> Original issue's description:
> > Moving MediaStreamSignaling logic into PeerConnection.
> >
> > This needs to happen because in the future, m-lines will be offered
> > based on the set of RtpSenders/RtpReceivers, rather than the set of
> > tracks that MediaStreamSignaling knows about.
> >
> > Besides that, MediaStreamSignaling was a "glue class" without
> > a clearly defined role, so it going away is good for other
> > reasons as well.
> >
> > Committed: https://crrev.com/97c392935411398b506861601c82e31d95c591f0
> > Cr-Commit-Position: refs/heads/master@{#10268}
>
> TBR=pthatcher@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/fc648b6d934e936f4d9a32c813364b331536ec3b
> Cr-Commit-Position: refs/heads/master@{#10269}
TBR=pthatcher@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1404473005
Cr-Commit-Position: refs/heads/master@{#10277}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| fc648b6d934e936f4d9a32c813364b331536ec3b |
|
14-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
Revert of Moving MediaStreamSignaling logic into PeerConnection. (patchset #10 id:180001 of https://codereview.webrtc.org/1393563002/ )
Reason for revert:
Broke browser_tests on Mac. Still need to investigate the cause.
Original issue's description:
> Moving MediaStreamSignaling logic into PeerConnection.
>
> This needs to happen because in the future, m-lines will be offered
> based on the set of RtpSenders/RtpReceivers, rather than the set of
> tracks that MediaStreamSignaling knows about.
>
> Besides that, MediaStreamSignaling was a "glue class" without
> a clearly defined role, so it going away is good for other
> reasons as well.
>
> Committed: https://crrev.com/97c392935411398b506861601c82e31d95c591f0
> Cr-Commit-Position: refs/heads/master@{#10268}
TBR=pthatcher@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1403633005
Cr-Commit-Position: refs/heads/master@{#10269}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 97c392935411398b506861601c82e31d95c591f0 |
|
13-Oct-2015 |
deadbeef <deadbeef@webrtc.org> |
Moving MediaStreamSignaling logic into PeerConnection.
This needs to happen because in the future, m-lines will be offered
based on the set of RtpSenders/RtpReceivers, rather than the set of
tracks that MediaStreamSignaling knows about.
Besides that, MediaStreamSignaling was a "glue class" without
a clearly defined role, so it going away is good for other
reasons as well.
Review URL: https://codereview.webrtc.org/1393563002
Cr-Commit-Position: refs/heads/master@{#10268}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| d4cec0d8fa7913bc9dfa9137e44cca9098e16698 |
|
09-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Remove MediaChannel::SetRemoteRenderer().
This is following discussion in: https://codereview.webrtc.org/1385893002/diff/60001/talk/media/webrtc/webrtcvoiceengine.cc#newcode2410
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1398823003
Cr-Commit-Position: refs/heads/master@{#10237}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 98c68865e715f693390209adb454ab3a5b6de332 |
|
09-Oct-2015 |
solenberg <solenberg@webrtc.org> |
- Remove AudioTrackRenderer.
- Remove AddChannel/RemoveChannel from AudioRenderer interface.
BUG=webrtc:4690
Committed: https://crrev.com/1c0bb386b67835feb5934f503dddfe0912bce3ac
Cr-Commit-Position: refs/heads/master@{#10226}
Review URL: https://codereview.webrtc.org/1399553003
Cr-Commit-Position: refs/heads/master@{#10235}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 4bac9c53da9988741d59753c2d789adb94de5e68 |
|
09-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Change SetOutputScaling to set a single level, not left/right levels.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1397773002
Cr-Commit-Position: refs/heads/master@{#10234}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| eefbc3bbd7a6962265b028cf259b5028944561d1 |
|
08-Oct-2015 |
torbjorng <torbjorng@webrtc.org> |
Revert of Remove AudioTrackRenderer (patchset #3 id:40001 of https://codereview.webrtc.org/1399553003/ )
Reason for revert:
Breaks Chrome since its build files were not updated prior to file removal.
Original issue's description:
> - Remove AudioTrackRenderer.
> - Remove AddChannel/RemoveChannel from AudioRenderer interface.
>
> BUG=webrtc:4690
>
> Committed: https://crrev.com/1c0bb386b67835feb5934f503dddfe0912bce3ac
> Cr-Commit-Position: refs/heads/master@{#10226}
TBR=tommi@webrtc.org,solenberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1393343003
Cr-Commit-Position: refs/heads/master@{#10228}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 1c0bb386b67835feb5934f503dddfe0912bce3ac |
|
08-Oct-2015 |
solenberg <solenberg@webrtc.org> |
- Remove AudioTrackRenderer.
- Remove AddChannel/RemoveChannel from AudioRenderer interface.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1399553003
Cr-Commit-Position: refs/heads/master@{#10226}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 335204c550e9570d356d0d6264475ac40c7f92f6 |
|
08-Oct-2015 |
torbjorng <torbjorng@webrtc.org> |
Revert of Provide RSA2048 as per RFC (patchset #9 id:200001 of https://codereview.webrtc.org/1329493005/ )
Reason for revert:
Breaks chrome.
Original issue's description:
> provide RSA2048 as per RFC
>
> BUG=webrtc:4972
>
> Committed: https://crrev.com/0df3eb03c9a6a8299d7e18c8c314ca58c2f0681e
> Cr-Commit-Position: refs/heads/master@{#10209}
TBR=hbos@webrtc.org,juberti@google.com,jbauch@webrtc.org,henrikg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4972
Review URL: https://codereview.webrtc.org/1397703002
Cr-Commit-Position: refs/heads/master@{#10210}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 0df3eb03c9a6a8299d7e18c8c314ca58c2f0681e |
|
08-Oct-2015 |
torbjorng <torbjorng@webrtc.org> |
provide RSA2048 as per RFC
BUG=webrtc:4972
Review URL: https://codereview.webrtc.org/1329493005
Cr-Commit-Position: refs/heads/master@{#10209}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 0c4e06b4c6107a1b94f764e279e4fb4161e905b0 |
|
07-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Use suffixed {uint,int}{8,16,32,64}_t types.
Removes the use of uint8, etc. in favor of uint8_t.
BUG=webrtc:5024
R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1362503003 .
Cr-Commit-Position: refs/heads/master@{#10196}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| facbbecb516547adc2ac684c8e0be95ad79dfd88 |
|
24-Sep-2015 |
solenberg <solenberg@webrtc.org> |
Remove use of DeviceManager from ChannelManager.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1346153002
Cr-Commit-Position: refs/heads/master@{#10042}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| cbecd358e032021eac11fb13e04ec7f070d4f407 |
|
23-Sep-2015 |
deadbeef <deadbeef@webrtc.org> |
Reland of TransportController refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/1358413003/ )
Reason for revert:
This CL just landed: https://codereview.chromium.org/1323243006/
Which fixes the FYI bots for the original CL, and breaks them for this revert.
Original issue's description:
> Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ )
>
> Reason for revert:
> This CL causes problems with the WebRTC-in-Chromium FYI bots. Presumably it needs to be done in several steps, where removed files are emptied instead of removed in the first step.
>
> Original issue's description:
> > TransportController refactoring.
> >
> > Getting rid of TransportProxy, and in its place adding a
> > TransportController class which will facilitate access to and manage
> > the lifetimes of Transports. These Transports will now be accessed
> > solely from the worker thread, simplifying their implementation.
> >
> > This refactoring also pulls Transport-related code out of BaseSession.
> > Which means that BaseChannels will now rely on the TransportController
> > interface to create channels, rather than BaseSession.
> >
> > Committed: https://crrev.com/47ee2f3b9f33e8938948c482c921d4e13a3acd83
> > Cr-Commit-Position: refs/heads/master@{#10022}
>
> TBR=pthatcher@webrtc.org,deadbeef@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/a81a42f584baa0d93a4b93da9632415e8922450c
> Cr-Commit-Position: refs/heads/master@{#10024}
TBR=pthatcher@webrtc.org,torbjorng@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1361773005
Cr-Commit-Position: refs/heads/master@{#10036}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| a81a42f584baa0d93a4b93da9632415e8922450c |
|
23-Sep-2015 |
torbjorng <torbjorng@webrtc.org> |
Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ )
Reason for revert:
This CL causes problems with the WebRTC-in-Chromium FYI bots. Presumably it needs to be done in several steps, where removed files are emptied instead of removed in the first step.
Original issue's description:
> TransportController refactoring.
>
> Getting rid of TransportProxy, and in its place adding a
> TransportController class which will facilitate access to and manage
> the lifetimes of Transports. These Transports will now be accessed
> solely from the worker thread, simplifying their implementation.
>
> This refactoring also pulls Transport-related code out of BaseSession.
> Which means that BaseChannels will now rely on the TransportController
> interface to create channels, rather than BaseSession.
>
> Committed: https://crrev.com/47ee2f3b9f33e8938948c482c921d4e13a3acd83
> Cr-Commit-Position: refs/heads/master@{#10022}
TBR=pthatcher@webrtc.org,deadbeef@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1358413003
Cr-Commit-Position: refs/heads/master@{#10024}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 47ee2f3b9f33e8938948c482c921d4e13a3acd83 |
|
23-Sep-2015 |
deadbeef <deadbeef@webrtc.org> |
TransportController refactoring.
Getting rid of TransportProxy, and in its place adding a
TransportController class which will facilitate access to and manage
the lifetimes of Transports. These Transports will now be accessed
solely from the worker thread, simplifying their implementation.
This refactoring also pulls Transport-related code out of BaseSession.
Which means that BaseChannels will now rely on the TransportController
interface to create channels, rather than BaseSession.
Review URL: https://codereview.webrtc.org/1350523003
Cr-Commit-Position: refs/heads/master@{#10022}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 8902433a43bbc9cc0de4966774d3dbbe37ef96fb |
|
18-Sep-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Revert "TransportController refactoring."
This reverts commit 9af63f473e1d0d6c47a741a046c41642dfc1c178.
Cr-Commit-Position: refs/heads/master@{#9994}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 9af63f473e1d0d6c47a741a046c41642dfc1c178 |
|
18-Sep-2015 |
deadbeef <deadbeef@webrtc.org> |
TransportController refactoring.
Getting rid of TransportProxy, and in its place adding a
TransportController class which will facilitate access to and manage
the lifetimes of Transports. These Transports will now be accessed
solely from the worker thread, simplifying their implementation.
This refactoring also pulls Transport-related code out of BaseSession.
Which means that BaseChannels will now rely on the TransportController
interface to create channels, rather than BaseSession.
This CL also adds some unit tests, and does some renaming.
For example, from "CandidateReady" to "CandidateGathered".
Review URL: https://codereview.webrtc.org/1246913005
Cr-Commit-Position: refs/heads/master@{#9993}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 7cbd188c5ed7df80bb737bd4ada94422730e2d89 |
|
18-Sep-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Remove GICE (again).
R=guoweis@webrtc.org
Review URL: https://codereview.webrtc.org/1353713002 .
Cr-Commit-Position: refs/heads/master@{#9979}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 91d6edef35e7275879c30ce16ecb8b6dc73c6e4a |
|
17-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Add RTC_ prefix to (D)CHECKs and related macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1335923002
Cr-Commit-Position: refs/heads/master@{#9964}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| d12140a68efdcffa1c2c18f25149905e9dae1a9c |
|
10-Sep-2015 |
guoweis <guoweis@webrtc.org> |
Revert change which removes GICE.
There are still dependencies on this functionality.
TBR=pthatcher@webrtc.org
BUG=526399
Review URL: https://codereview.webrtc.org/1336553003
Cr-Commit-Position: refs/heads/master@{#9920}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 66f43392a31ac566565e910246ef496fcbbafb04 |
|
09-Sep-2015 |
solenberg <solenberg@webrtc.org> |
Remove [Voice|Video]MediaChannel::GetOptions().
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1324853003
Cr-Commit-Position: refs/heads/master@{#9904}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 3cc834ae8628ef042497d8effe4bd223235bcd28 |
|
05-Sep-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Add more IceCandidatePairType for host-host CandidatePair
This is to help to differentiate endpoints which are behind NAT or on the public internet.
BUG=520101
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1328453003 .
Cr-Commit-Position: refs/heads/master@{#9864}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 3a14bf311f366602ebc72314ca8906be61a70da4 |
|
31-Aug-2015 |
Henrik Boström <hbos@webrtc.org> |
Replacing SSLIdentity* with scoped_refptr<RTCCertificate> in the cricket::TransportDescriptionFactory layers.
Updates TransportDescriptionFactory, calls and unittests.
BUG=webrtc:4927
R=tommi@webrtc.org, torbjorng@webrtc.org
Review URL: https://codereview.webrtc.org/1311903004 .
Cr-Commit-Position: refs/heads/master@{#9815}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| d82819892a382899a82ced756a9922a84ca9ca98 |
|
27-Aug-2015 |
Henrik Boström <hbos@webrtc.org> |
Replaces SSLIdentity* with scoped_refptr<RTCCertificate> in the cricket::Transport layer.
Why the replacements? Mainly two reasons:
1) RTCCertificate owns the identity and as long as things are referencing the identity there should be a scoped_refptr reference to the RTCCertificate. Handing out raw pointers is less memory safe.
2) With the latest RFC, an RTCCertificate should be sufficient for specifying a crypto cert and the code should be updated to use RTCCertificate instead of SSLIdentity directly.
This replace work is split up into multiple CLs. In this CL...
- WebRtcSessionDescriptionFactory is updated to use RTCCertificate over SSLIdentity.
- WebRtcSessionDescriptionFactory::SignalCertificateReady is connected to WebRtcSession::OnCertificateReady and WebRtcSession is updated to use RTCCertificate.
- The cricket::Transport and related classes are updated to use RTCCertificate. These are called from WebRtcSession::OnCertificateReady.
BUG=webrtc:4927
R=tommi@webrtc.org, torbjorng@webrtc.org
Review URL: https://codereview.webrtc.org/1312643004 .
Cr-Commit-Position: refs/heads/master@{#9794}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 87713d0fe6fb9c86abe501bdf3d26ef4287ee617 |
|
25-Aug-2015 |
Henrik Boström <hbos@webrtc.org> |
RTCCertificates added to RTCConfiguration, used by WebRtcSession/-DescriptionFactory.
This CL allows you to, having generated one or more RTCCertificates, supply them to RTCConfiguration for CreatePeerConnection use. This means an SSLIdentity does not have to be generated with a DtlsIdentityStore[Interface/Impl] as part of the CreatePeerConnection steps because the certificate contains all the necessary information.
To create an RTCCertificate you have to do the identity generation yourself though. But you could reuse the same RTCCertificate for multiple connections.
BUG=webrtc:4927
R=tommi@webrtc.org, torbjorng@webrtc.org
Review URL: https://codereview.webrtc.org/1288033009 .
Cr-Commit-Position: refs/heads/master@{#9774}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 2159b89fa2cb55beeef38f72bd45e217f3d33d4e |
|
22-Aug-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots.
This reverts commit 5bdafd44c86ee46bd7e040f19828324583418b33.
Original CL: https://codereview.webrtc.org/1263663002/
R=guoweis@webrtc.org
Review URL: https://codereview.webrtc.org/1303393002 .
Cr-Commit-Position: refs/heads/master@{#9761}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 5bdafd44c86ee46bd7e040f19828324583418b33 |
|
21-Aug-2015 |
minyuel <minyue@webrtc.org> |
Revert "Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots.""
This reverts commit 081f34b564e1a26ffbbe9515eba1fef7c736fdde.
Original code review see
https://codereview.webrtc.org/1291363005
The revert is due to a suspicion of "Reland "Remove GICE..." being the cause of failure on Linux memcheck, see
https://build.chromium.org/p/client.webrtc/builders/Linux%20Memcheck/builds/4137
TBR=pthatcher@webrtc.org,
BUG=
Review URL: https://codereview.webrtc.org/1308753003 .
Cr-Commit-Position: refs/heads/master@{#9756}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 081f34b564e1a26ffbbe9515eba1fef7c736fdde |
|
20-Aug-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots."
This reverts commit 475243a134be003aab30bb17294ca6c664d0ef81.
R=guoweis@webrtc.org
Review URL: https://codereview.webrtc.org/1291363005 .
Cr-Commit-Position: refs/heads/master@{#9738}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 3d564c10157d7de1d2d4236f4e2a13ff1363d52b |
|
20-Aug-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Add instrumentation to track the IceEndpointType.
The IceEndpointType has the format of <local_endpoint>_<remote_endpoint>. It is recorded on the BestConnection when we have the first OnTransportCompleted signaled.
BUG=webrtc:4918
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1277263002 .
Cr-Commit-Position: refs/heads/master@{#9737}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| b6d4ec418504fd947c6f96829c73180e9487e203 |
|
17-Aug-2015 |
Torbjorn Granlund <torbjorng@google.com> |
Support generation of EC keys using P256 curve and support ECDSA certs.
This CL started life here: https://webrtc-codereview.appspot.com/51189004
BUG=webrtc:4685, webrtc:4686
R=hbos@webrtc.org, juberti@webrtc.org
Review URL: https://codereview.webrtc.org/1189583002 .
Cr-Commit-Position: refs/heads/master@{#9718}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| fa301809b698017455847f45cc7e0dfa1bdfed35 |
|
11-Aug-2015 |
pthatcher <pthatcher@webrtc.org> |
Revert "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots.
This reverts commit 3449faa553ec94c52ef2d0949867befb60992c88.
TBR=deadbeef@webrtc.org, juberti@webrtc.org
NOPRESUBMIT=true
Review URL: https://codereview.webrtc.org/1274273005
Cr-Commit-Position: refs/heads/master@{#9698}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 5e56c5927e097f095aef2e9f7be49fd3d59221e1 |
|
11-Aug-2015 |
Henrik Boström <hbos@webrtc.org> |
DtlsIdentityStoreInterface added and the implementation is called DtlsIdentityStoreImpl (previously named without the -Impl bit and without an interface).
DtlsIdentityStoreImpl is updated to take KeyType into account, something which will be relevant after this CL lands:
https://codereview.webrtc.org/1189583002
The DtlsIdentityService[Interface] classes are about to be removed (to be removed when Chromium no longer implements and uses the interface). This was an unnecessary layer of complexity. The FakeIdentityService is now instead a FakeDtlsIdentityStore.
Where a service was previously passed around, a store is now passed around.
Identity generation is now commonly performed using DtlsIdentityStoreInterface. Previously, if a service was not specified, WebRtcSessionDescriptionFactory could fall back on its own generation code. Now, a store has to be provided for generation to occur.
For more information about the steps being taken to land this without breaking Chromium, see referenced bug.
BUG=webrtc:4899
R=magjed@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1176383004 .
Cr-Commit-Position: refs/heads/master@{#9696}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 3449faa553ec94c52ef2d0949867befb60992c88 |
|
10-Aug-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever).
R=deadbeef@webrtc.org, juberti@webrtc.org
Review URL: https://codereview.webrtc.org/1263663002 .
Cr-Commit-Position: refs/heads/master@{#9692}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 503726c3498201822079c5abe9e528498846c9f2 |
|
31-Jul-2015 |
honghaiz <honghaiz@webrtc.org> |
Fix the generation mismatch assertion error.
BUG=4860
Review URL: https://codereview.webrtc.org/1248063002
Cr-Commit-Position: refs/heads/master@{#9667}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| fabe2c961f9cf86d519532a96e96fa7d6c4ca37d |
|
16-Jul-2015 |
jbauch <jbauch@webrtc.org> |
Remove deprecated functions.
This CL removes some functions that are marked as deprecated. Chromium
has been updated in https://crrev.com/7dee3f68b7699ad72c7fc4d75332f72703313849
to call the new functions.
Review URL: https://codereview.webrtc.org/1237613003
Cr-Commit-Position: refs/heads/master@{#9598}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| f39382943449b7e44ac563e05a14203534591acf |
|
15-Jul-2015 |
deadbeef <deadbeef@webrtc.org> |
Use "UDP/TLS/RTP/SAVPF" profile in offer when DTLS-SRTP is used.
Tested that this doesn't break compatibility with Firefox or older
versions of Chrome, no matter which side generates the initial offer.
BUG=webrtc:2796
Review URL: https://codereview.webrtc.org/1219333002
Cr-Commit-Position: refs/heads/master@{#9589}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 0f620f4e318a162e7a251133e7a8ddea5188b9bb |
|
09-Jul-2015 |
tommi <tommi@webrtc.org> |
Make sure we process all pending offer/answer requests before terminating.
This fixes a bug in the WebRtcSessionDescriptionFactory where messages would be dropped or worse yet processed after the factory was deleted.
BUG=chromium:507307
Review URL: https://codereview.webrtc.org/1231823002
Cr-Commit-Position: refs/heads/master@{#9557}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| ac8869ec5a606e0a0ab71e70937c8fbf403630ce |
|
03-Jul-2015 |
jbauch <jbauch@webrtc.org> |
Report metrics about negotiated ciphers.
This CL adds an API to the metrics observer interface to report negotiated
ciphers for WebRTC sessions. This can be used from Chromium for UMA metrics
later to get an idea which cipher suites are used by clients (e.g. compare
the use of DTLS 1.0 / 1.2).
BUG=428343
Review URL: https://codereview.webrtc.org/1156143005
Cr-Commit-Position: refs/heads/master@{#9537}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| d4f769d8fc48769eff226392b1ae105161b3e7c4 |
|
28-May-2015 |
Donald Curtis <decurtis@webrtc.org> |
Stop video candidates getting down to audio.
Second attempt at adding a check to make sure that the video
transportproxy doesn't send down candidates to the audio transport
channel when things are bundled.
BUG=4665
R=juberti@google.com, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50059004
Cr-Commit-Position: refs/heads/master@{#9316}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 4bf12eafba4e18504ac22080262dba61b4c8b9e7 |
|
23-May-2015 |
Alejandro Luebs <aluebs@webrtc.org> |
Revert "Fix sending wrong candidates down to transportchannel."
This reverts commit f65de8483e90d1d52d5d8f40f646e77bf45b10ea.
It was breaking the build bots: http://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Light/builds/3062
TBR=decurtis
BUG=
Review URL: https://webrtc-codereview.appspot.com/54539004
Cr-Commit-Position: refs/heads/master@{#9267}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| f65de8483e90d1d52d5d8f40f646e77bf45b10ea |
|
22-May-2015 |
Donald Curtis <decurtis@webrtc.org> |
Fix sending wrong candidates down to transportchannel.
BUG=4665
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54489004
Cr-Commit-Position: refs/heads/master@{#9266}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| af55ccc054de9b91f6e5f5059937a91c0c91ff30 |
|
21-May-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Add RtcpMuxPolicy support to PeerConnection.
BUG=4611
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/46169004
Cr-Commit-Position: refs/heads/master@{#9251}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 64dad838e61e92e4a72437b153c5eba7a200fb4a |
|
11-May-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
The original change was reverted due to a breakage in the chrome build.
This change includes a fix for this.
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49329004
Cr-Commit-Position: refs/heads/master@{#9169}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 1f629232d5f852452499104c28e7d61c7b0b8c77 |
|
10-May-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Revert r9164 "Adding a new constraint to set NetEq buffer capacity ..."
This reverts commit fd32f35aff8fc28ec084bddc274de284e0422a57.
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/55369004
Cr-Commit-Position: refs/heads/master@{#9165}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| fd32f35aff8fc28ec084bddc274de284e0422a57 |
|
10-May-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
This reverts commit cdb47a4533b7b1e29e803ed6591a68bb1a4f1692.
Contains a tentative fix to the chrome build breakage caused by the
original change.
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47139004
Cr-Commit-Position: refs/heads/master@{#9164}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| cdb47a4533b7b1e29e803ed6591a68bb1a4f1692 |
|
08-May-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Revert r9159 "Adding a new constraint to set NetEq buffer capacity ..."
This reverts commit 208a2294cde839025318f1b3d57559cb0611a4e7.
Breaks the Chrome build.
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/53399004
Cr-Commit-Position: refs/heads/master@{#9161}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 208a2294cde839025318f1b3d57559cb0611a4e7 |
|
08-May-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Adding a new constraint to set NetEq buffer capacity from peerconnection
This change makes it possible to set a custom value for the maximum
capacity of the packet buffer in NetEq (the audio jitter buffer). The
default value is 50 packets, but any value can be set with the new
functionality.
R=jmarusic@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50869004
Cr-Commit-Position: refs/heads/master@{#9159}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 4eddf18b1c4a93bc9c736783094cc4204c2d955e |
|
30-Apr-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Don't crash if SetRemoteDescription is called first with BundlePolicy=max-bundle.
BUG=
R=decurtis@webrtc.org, juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/46149004
Cr-Commit-Position: refs/heads/master@{#9124}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 0e209b03bf55d6daf209e35b3a8e8b6eab3d4d52 |
|
24-Mar-2015 |
Donald Curtis <decurtis@webrtc.org> |
Update bundle behavior to match BundlePolicy spec in http://rtcweb-wg.github.io/jsep/.
BUG=1574
R=juberti@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36659004
Cr-Commit-Position: refs/heads/master@{#8851}
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 2d25b44f470afdd56513b75d641166f6e7cdcd04 |
|
16-Mar-2015 |
changbin.shao@webrtc.org <changbin.shao@webrtc.org> |
Check associated payload type when negotiate RTX codecs.
At the moment, only payload name is checked when match two RTX codecs.
This will cause wrong behavior of codec negotiation if multiple RTX codecs
are added.
BUG=
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34189004
Cr-Commit-Position: refs/heads/master@{#8727}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8727 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| b4aac13810815f77b019f9db9d0300862c8313bc |
|
13-Mar-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Cleanup SocketMonitor a little so that it can handle a change in transport channel. And cleanup some names and style and such as well.
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/
R=guoweis@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49399004
Cr-Commit-Position: refs/heads/master@{#8720}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8720 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 4f85288e71136671ae194fcdd730e2d0f0241db9 |
|
12-Mar-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Socket options are only applied when first setting TransportChannelImpl.
Also fixed the issue when we have an TransportChannelImpl, the socket
option is not preserved.
Since this is a code path that will be modified by bundle (which Peter also has a test case already), we don't need a test case here.
BUG=4374
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42699004
Cr-Commit-Position: refs/heads/master@{#8702}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8702 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 14665ff7d4024d07e58622f498b23fd980001871 |
|
04-Mar-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
Clang version changed 223108:230914
Details: https://chromium.googlesource.com/chromium/src/+/e144d30..6fdb142/tools/clang/scripts/update.sh
Removes the OVERRIDE macro defined in:
* webrtc/base/common.h
* webrtc/typedefs.h
The majority of the source changes were done by running this in src/:
perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h" -o -name "*.cc*" -o -name "*.mm*"`
which converted all:
virtual Foo() OVERRIDE
functions to:
Foo() override
Then I manually edited:
* talk/media/webrtc/fakewebrtccommon.h
* webrtc/test/fake_common.h
Remaining uses of OVERRIDE was fixed by search+replace.
Manual edits were done to fix virtual destructors that were
overriding inherited ones.
Finally a build error related to the pure virtual definitions of
Read, Write and Rewind in common_types.h required a bit of
refactoring in:
* webrtc/common_types.cc
* webrtc/common_types.h
* webrtc/system_wrappers/interface/file_wrapper.h
* webrtc/system_wrappers/source/file_impl.cc
This roll should make it possible for us to finally re-enable deadlock
detection for TSan on the buildbots.
BUG=4106
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41069004
Cr-Commit-Position: refs/heads/master@{#8596}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 804eb468066bc930cf862652868481740dfaad95 |
|
20-Feb-2015 |
jlmiller@webrtc.org <jlmiller@webrtc.org> |
Change default from GICE to ICE5245 for SDP offers
BUG=4299
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34289004
Cr-Commit-Position: refs/heads/master@{#8440}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8440 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 877ac765ad30a22148da41695fa607682af4a191 |
|
04-Feb-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Cleanup and prepare for bundling.
- Add a GetOptions function. Needed for eventual bundle testing to
confirm that channel options are preserved.
- Simplify unit tests and cleanup unused code.
This is a re-roll of 8237 (https://webrtc-codereview.appspot.com/39699004) with a default GetOption implementation.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38909004
Cr-Commit-Position: refs/heads/master@{#8245}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8245 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| c5f697135e626044b15eacdc82fd840fbe74b351 |
|
04-Feb-2015 |
bjornv@webrtc.org <bjornv@webrtc.org> |
Revert 8237 "Cleanup and prepare for bundling."
libjingle_peerconnection_objc_test consistently failing on Mac64 Debug.
> Cleanup and prepare for bundling.
>
> - Add a GetOptions function. Needed for eventual bundle testing to
> confirm that channel options are preserved.
> - Simplify unit tests and cleanup unused code.
>
> BUG=1574
> R=pthatcher@webrtc.org, tommi@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/39699004
TBR=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34959004
Cr-Commit-Position: refs/heads/master@{#8241}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8241 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| af01d93aa2d75b39cdcaadd682c5c60336c75ea7 |
|
04-Feb-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Cleanup and prepare for bundling.
- Add a GetOptions function. Needed for eventual bundle testing to
confirm that channel options are preserved.
- Simplify unit tests and cleanup unused code.
BUG=1574
R=pthatcher@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39699004
Cr-Commit-Position: refs/heads/master@{#8237}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8237 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| dacdd9403d30cdb13ab2de645841edd2ae76950d |
|
23-Jan-2015 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Reland r7980:
Accept incoming pings before remote answer is set, to reduce connection latency.
Set ICE connection state to 'checking' after setting the remote answer, so that it can transition into 'connected' if the peer reflexive connection is up before any remote candidate is set. See more details in crbug/446908
BUG=4068, crbug/446908
R=juberti@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8141 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 5f93d0a140515e3b8cdd1b9a4c6f5871144e5dee |
|
20-Jan-2015 |
jlmiller@webrtc.org <jlmiller@webrtc.org> |
Update libjingle license statements at top of talk files for consistency
BUG=2133
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8105 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 9657265f391cfe473a61b18a4579bbbeb44c9bd8 |
|
09-Jan-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Revert "Accept incoming pings before remote answer is set to reduce connection latency."
This reverts r7980.
It was causing the ICE connected state to happen while still in the new state rather than going through the checking state, which was causing an ASSERT to fire, which was causing a crash.
Review URL: https://webrtc-codereview.appspot.com/41429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8031 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| c5fd66dcdfdba3ec114cc5b5c0337eba503cee40 |
|
29-Dec-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Accept incoming pings before remote answer is set to reduce connection latency.
BUG=4068
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7980 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 7169afd9d53fce803858bc954e6cc5ebbf9b1695 |
|
04-Dec-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
With IPv6 enabled, it's important to know whether IPv6 is really used or not. BestConnection is tracked for this purpose. Also added a test case to verify the end to end behavior.
BUG=411086
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30919005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7814 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 269fb4bc90b79bebbb8311da0110ccd6803fd0a8 |
|
28-Oct-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
move xmpp and p2p to webrtc
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and
webrtc/p2p. Also makes libjingle use those version instead of the one in the talk folder.
BUG=3379
Review URL: https://webrtc-codereview.appspot.com/26999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7549 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 28100cb38896fe298b6df11ffd31838d9faf5b8a |
|
18-Oct-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p."
BUG=N/A
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7472 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| d1ba6d9cbfc44618d2c553ff7851948c730ae37b |
|
15-Oct-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p.
BUG=3379
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27709005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7459 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 742922b313baaebfbacf735287f9729a8bc6f8e0 |
|
07-Oct-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Make the media content send only if offerToReceive is false while local streams exist.
We previously do not add the media content if offerToReceive is false.
BUG=3833
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7390 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 34f2a9ea7245bac103fececfa53e92359680467a |
|
28-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Initialize SSL in unittest_main.cc.
Instead of having each test individually initialize and tear down SSL
move this to unittest_main.cc so that all tests are properly
initialized and new tests "don't have to think about it".
R=pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/30549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7316 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| bebc75e8bdd4172fec69ee376634ecbeb1191992 |
|
27-Sep-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Fix the duplicated candidate problem when using multiple STUN servers.
BUG=3723
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7312 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 7d4891d3f18861bdd5ec5d27409110cf3d110fa1 |
|
09-Sep-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Fixes two issues in how we handle OfferToReceiveX for CreateOffer:
1. the options set in the first CreateOffer call should not affect the result of a second CreateOffer call, if SetLocalDescription is not called after the first CreateOffer. So the member var options_ of MediaStreamSignaling is removed to make each CreateOffer independent.
Instead, MediaSession is responsible to make sure that an m-line in the current local description is never removed from the newly created offer.
2. OfferToReceiveAudio used to default to true, i.e. always having m=audio line even if no audio track. This is changed so that by default m=audio is only added if there are any audio tracks.
BUG=2108
R=pthatcher@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7068
Review URL: https://webrtc-codereview.appspot.com/16309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7124 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 3d81b1b22a3ddc2047e95e74ca28dffa2bbfdaae |
|
09-Sep-2014 |
mallinath@webrtc.org <mallinath@webrtc.org> |
Relanding https://code.google.com/p/webrtc/source/detail?r=7093, after it got
reverted due to some internal compile failures.
In this CL changes are done in portallocator_unittest.cc, in particular to EXPECT_EQ checking in new tests.
Original patch committed in https://code.google.com/p/webrtc/source/detail?r=7093
TBR=juberti@webrtc.org
BUG=1179
Review URL: https://webrtc-codereview.appspot.com/22329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7118 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 8b0b21161abdcdc2f2528aadf25f1f8f5c99e8b2 |
|
09-Sep-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Revert 7093: "Implementing ICE Transports type handling in libjingle transport."
TBR=mallinath@webrtc.org
BUG=N/A
Review URL: https://webrtc-codereview.appspot.com/28419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7112 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| c172320bd22311a0cf8c7c51c5c782e321622de1 |
|
08-Sep-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Revert "Fixes two issues in how we handle OfferToReceiveX for CreateOffer:" because it broke content_browsertests on Android.
This reverts commit r7068.
TBR=kjellander@webrtc.org
BUG=2108
Review URL: https://webrtc-codereview.appspot.com/23539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7108 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 7256d31d2864379d3299362f64b7a23741e67adb |
|
07-Sep-2014 |
mallinath@webrtc.org <mallinath@webrtc.org> |
Implementing ICE Transports type handling in libjingle transport.
BUG=1179
R=juberti@webrtc.org, bemasc@webrtc.org, jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7093 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 52055a276df3b0b0c3ed4c58ea74e0a4d8fe3891 |
|
04-Sep-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Fixes two issues in how we handle OfferToReceiveX for CreateOffer:
1. the options set in the first CreateOffer call should not affect the result of a second CreateOffer call, if SetLocalDescription is not called after the first CreateOffer. So the member var options_ of MediaStreamSignaling is removed to make each CreateOffer independent.
Instead, MediaSession is responsible to make sure that an m-line in the current local description is never removed from the newly created offer.
2. OfferToReceiveAudio used to default to true, i.e. always having m=audio line even if no audio track. This is changed so that by default m=audio is only added if there are any audio tracks.
BUG=2108
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7068 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| b4c7b09c1352174ecc1faf8c0cd93c66028a0485 |
|
25-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 73927775-> 74032598
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6965 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| a09a99950ec40aef6421e4ba35eee7196b7a6e68 |
|
13-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 73222930-> 73226398
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6891 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 53df88c1bcd42d79d178f8c8da8d4d620f1c12cf |
|
08-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72847605-> 72850595
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6855 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| b18bf5e47d1db8ca563c9c6f12e77f9cd63879d4 |
|
04-Aug-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds the support of RTCOfferOptions for PeerConnectionInterface::CreateOffer.
Constraints are still supported for CreateOffer, but converted to RTCOfferOptions internally.
BUG=3282
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6822 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| d4e598d57aed714a599444a7eab5e8fdde52a950 |
|
29-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72097588-> 72159069
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 51c5508bf1489f6b65bde2373b97cdf2e3af2426 |
|
29-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72016417-> 72097588
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6792 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 45304ff0a712cf23d0de98d9e8f4fc576971b120 |
|
24-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 71829282-> 71834788
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6773 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| e2da234e2767ad6e6433fbeee998b0f681100981 |
|
23-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 71766184-> 71775619
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6768 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| a0b929b63c35089a086715640149cdd24960fb2b |
|
19-Jul-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert "Reland r6707 with the fix for callclient.cc."
Breaking pulse build again.
This reverts commit 3e0bb9b5bf7f616000399e24f1d9622ad6b612f9.
TBR=wu@webrtc.org
BUG=3310
Review URL: https://webrtc-codereview.appspot.com/17979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6740 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| a6e8cf8fb72b3c5bf331938ddb86093559c1c631 |
|
18-Jul-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Reland r6707 with the fix for callclient.cc.
TBR=mallinath@webrtc.org
BUG=3310
Review URL: https://webrtc-codereview.appspot.com/13039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6737 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| e10d28cf14d55a86138da97cbf87ca06bb2f5589 |
|
17-Jul-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
fix
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6720 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 52eddec71b45e8e5ff1294040b8cb658dd144c7a |
|
17-Jul-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 6707 "Add support of multiple STUN servers in UDPPort."
Reason:
Breaks the build on callclient.cc.
> Add support of multiple STUN servers in UDPPort.
> Now UDPPort signals PortComplete or PortError when the Bind requests for all STUN servers are responded or failed. If any STUN bind is successful, PortComplete is signaled; otherwise, PortError is signaled.
>
> I discovered a bug in SocketAddress while working on this. It didn't consider two addresses unequal if they have unresolved IP and different hosts. It's fixed now.
>
> BUG=3310
> R=mallinath@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/13879004
TBR=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6711 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 46fb331bc5836eb03bc0cbda46097d9089a19561 |
|
16-Jul-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add support of multiple STUN servers in UDPPort.
Now UDPPort signals PortComplete or PortError when the Bind requests for all STUN servers are responded or failed. If any STUN bind is successful, PortComplete is signaled; otherwise, PortError is signaled.
I discovered a bug in SocketAddress while working on this. It didn't consider two addresses unequal if they have unresolved IP and different hosts. It's fixed now.
BUG=3310
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6707 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 174a67439b03cc9c98bcc7fb426ddda8855a0fc2 |
|
02-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Enable -Wall, -Wextra and -Wunused-variable for talk/ on clang.
Also removes one case of unused-variable.
BUG=3220
R=henrike@webrtc.org, kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15619005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6297 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 7aa1a4767f14b58924822a0b7b30b265870fa806 |
|
23-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 67848628-> 67848776
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6237 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 41451d4e55e9cc00c342d0ad64dcf891cfb24622 |
|
03-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66106643-> 66138442
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6049 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 4e393070be2288596170e4ac21783785ab511466 |
|
07-Apr-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Compare the answer's media type against offer to make sure they are match. Otherwise we should return failure.
BUG=2687
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11079005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5858 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 6e3dbc2a77eb96b050c4909c4206348f1b15550c |
|
25-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 63648983-> 63738002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5779 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| b90991dade9139e5c14c3b616a9eff07b9d6fdda |
|
04-Mar-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle 62472237->62550414
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5640 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| d3dc424fe5f330be273065fa1fee0ebca0f0771d |
|
01-Mar-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove posting of ICE messages from WebRTCSession in PeerConnection to signaling thread.
These callbacks are called from signal thread already. There is no point
in posting messages on the same thread again.
BUG=2922
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5626 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| a7b981843f35bb6c26cf3bc95b5a00a0b9f50a93 |
|
21-Feb-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Unrevert 5590 "description"(=(Auto)update libjingle 61834300->61901702).
BUG=N/A
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5595 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| ef2215110c00ee1d8225b08815bfdcee918767f9 |
|
21-Feb-2014 |
xians@webrtc.org <xians@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5590 "description"
> description
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8949006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5593 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 2643805a2057b92e916bcf4f71668bc80766625e |
|
20-Feb-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
description
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5590 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 385857dfd414dcc1fb4941218b52417808349030 |
|
14-Feb-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 61549749.
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5549 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 67ee6b9a6260fa80b83326c4b4fec8857c0e578c |
|
03-Feb-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 60923971
Review URL: https://webrtc-codereview.appspot.com/7909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5475 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 4b26e2eee3e3b2a0c22946372a38f7efa6cee146 |
|
16-Jan-2014 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 59676287
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5390 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| aebb1ade9d760841f243e380fa22b7ecff2d3ecc |
|
14-Jan-2014 |
henrika@webrtc.org <henrika@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
pRevert 5371 "Revert 5367 "Update talk to 59410372.""
> Revert 5367 "Update talk to 59410372."
>
> > Update talk to 59410372.
> >
> > R=jiayl@webrtc.org, wu@webrtc.org
> >
> > Review URL: https://webrtc-codereview.appspot.com/6929004
>
> TBR=mallinath@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/6999004
TBR=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5381 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 44461fa5cbecd556691b0ba963f95973f6abece1 |
|
13-Jan-2014 |
henrika@webrtc.org <henrika@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5367 "Update talk to 59410372."
> Update talk to 59410372.
>
> R=jiayl@webrtc.org, wu@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/6929004
TBR=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5371 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 0f3356e20b70416f13e12ef596da66f6c347eea7 |
|
11-Jan-2014 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 59410372.
R=jiayl@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5367 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 364f204d16d1f10cf01b1b5543ce020c3e9961b8 |
|
20-Nov-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 56698267.
TBR=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/4119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5143 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| de305014c62832a382d38144a9dc518cf1d02f88 |
|
31-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 55906045.
Review URL: https://webrtc-codereview.appspot.com/3159005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5065 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 97077a3ab27259164eb121034b6e0ebe9ba592df |
|
25-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 55618622.
Update libyuv to r826.
TEST=try bots
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5038 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 19f27e6a24f877fc2b0409a94b02d5f40ba3dc8c |
|
13-Oct-2013 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 54527154.
TBR=wu
Review URL: https://webrtc-codereview.appspot.com/2389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4954 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 7e809c323a37ddd06469c6df815e4eab6c15559a |
|
30-Sep-2013 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to CL 53496343.
Review URL: https://webrtc-codereview.appspot.com/2323005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4882 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| a27be8e4a1f59a51ecafba71ba30ddd0bcc9f1f1 |
|
28-Sep-2013 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to CL 53398036.
Review URL: https://webrtc-codereview.appspot.com/2323004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4872 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 1112c30e1e5f5c7b4b517c4954ef3f15b989a996 |
|
23-Sep-2013 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 53057474.
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2274004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4818 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 967bfff54d00f176a554bf9f955f14dde99f7bb9 |
|
19-Sep-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 52534915.
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/2251004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4786 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 8a1448950cc26dabe50105d7af6b37e8ca93a233 |
|
14-Sep-2013 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Disable WebRtcSessionTest.TestCreateOfferWithSctpEnabledWithoutStreams
BUG=2374
TBR=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2214004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4747 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| a59696b2a5f0c138d4176249bac223ad6c4316d5 |
|
14-Sep-2013 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 52300956
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2213004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4744 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| cadf9040cbb9e7bb1b73a95e43e7d228fe6b2bdb |
|
30-Aug-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 51664136.
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2148004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4649 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 7666db79fa269c6688651008edd8cf88276c0671 |
|
22-Aug-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 51242664.
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2090005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4594 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 822fbd8b68ffdb481b9557e2950ae8d6657c8ce6 |
|
16-Aug-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 50918584.
Together with Stefan's http://review.webrtc.org/1960004/.
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2048004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| a5506690b408794a122eee6d06ebebb75a2d4287 |
|
12-Aug-2013 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 50733053.
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2017004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4532 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 91053e7c5a743f4a92f5079844b0747c927f3bbd |
|
10-Aug-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 50654631.
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2000006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4519 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 1e09a711263dd105e6f7a03812250084c64e5fd8 |
|
26-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk folder to revision=49952949
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4413 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 28654cbc2256230c978f41cbaf550bc2e9c2f2db |
|
22-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk folder to revision=49713299.
TBR=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1848004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4380 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 9de257d00f1f805af28f15fd814a8a84460028e5 |
|
17-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk folder to revision=49470012. Same as 375 in libjingle's google code repository.
TBR=wu@webrtc.org
BUG=N/A
Review URL: https://webrtc-codereview.appspot.com/1824004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4364 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
|
| 723d683ecbe6a934885a60712c66ca2c01700a51 |
|
12-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk folder to revision=49260075. Same as 369 in libjingle's google code repository.
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1797004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4338 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
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| 28e20752806a492f5a6a5d343c02f9556f39b1cd |
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10-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds trunk/talk folder of revision 359 from libjingles google code to
trunk/talk
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/webrtcsession_unittest.cc
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