| 8ac544e811439f79b2ec0c676f383ddc51ef2ed5 |
|
08-Oct-2015 |
tfarina <tfarina@chromium.org> |
Get rid of deprecated SocketAddress::IsAny() method.
This patch converts the usage of IsAny() to IsAnyIP() and removes the
deprecated method.
BUG=None
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1392153002
Cr-Commit-Position: refs/heads/master@{#10220}
/external/webrtc/webrtc/base/virtualsocketserver.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/webrtc/base/virtualsocketserver.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/webrtc/base/virtualsocketserver.cc
|
| 384194369b4be41912353631a68689129a49e58c |
|
16-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Consolidate constructormagic macros with Chromium version and remove Chromium override.
Part of work removing dependency on Chromium's base.
Only adds "= delete". From https://codereview.chromium.org/1151443003 :
"This will guarantee the error to be at compile time, and not rely on the call visibility (private)."
In consequence of that change, fixed an illegal copy and removed a bunch of unused variables.
Depends on https://codereview.webrtc.org/1345433002/
BUG=chromium:468375
(in particular comment #37)
NOTRY=true
Review URL: https://codereview.webrtc.org/1342543004
Cr-Commit-Position: refs/heads/master@{#9954}
/external/webrtc/webrtc/base/virtualsocketserver.cc
|
| 9a78d22822880884f9fa495e4cbe33f5224296c4 |
|
10-Sep-2015 |
tommi <tommi@webrtc.org> |
Revert of Consolidate constructormagic macros with Chromium version and remove Chromium override. (patchset #4 id:60001 of https://codereview.webrtc.org/1316363005/ )
Reason for revert:
Had to revert since FYI bots stopped compiling. Example failure:
[94/9470] CXX obj\third_party\webrtc\modules\video_processing\main\source\video_processing_sse2.content_analysis_sse2.obj
FAILED: ninja -t msvc -e environment.x86 -- E:\b\build\goma/gomacc "E:\b\depot_tools\win_toolchain\vs2013_files\VC\bin\amd64_x86\cl.exe" /nologo /showIncludes /FC @obj\third_party\webrtc\modules\video_coding\codecs\h264\webrtc_h264.h264.obj.rsp /c ..\..\third_party\webrtc\modules\video_coding\codecs\h264\h264.cc /Foobj\third_party\webrtc\modules\video_coding\codecs\h264\webrtc_h264.h264.obj /Fdobj\third_party\webrtc\modules\webrtc_h264.cc.pdb
e:\b\build\slave\win\build\src\base\macros.h(28) : error C2220: warning treated as error - no 'object' file generated
e:\b\build\slave\win\build\src\base\macros.h(28) : warning C4005: 'DISALLOW_COPY_AND_ASSIGN' : macro redefinition
e:\b\build\slave\win\build\src\third_party\webrtc\base\constructormagic.h(27) : see previous definition of 'DISALLOW_COPY_AND_ASSIGN'
FAILED: ninja -t msvc -e environment.x86 -- E:\b\build\goma/gomacc "E:\b\depot_tools\win_toolchain\vs2013_files\VC\bin\amd64_x86\cl.exe" /nologo /showIncludes /FC @obj\third_party\webrtc\base\rtc_base_approved.bitbuffer.obj.rsp /c ..\..\third_party\webrtc\base\bitbuffer.cc /Foobj\third_party\webrtc\base\rtc_base_approved.bitbuffer.obj /Fdobj\third_party\webrtc\base\rtc_base_approved.cc.pdb
e:\b\build\slave\win\build\src\base\macros.h(28) : error C2220: warning treated as error - no 'object' file generated
e:\b\build\slave\win\build\src\base\macros.h(28) : warning C4005: 'DISALLOW_COPY_AND_ASSIGN' : macro redefinition
e:\b\build\slave\win\build\src\third_party\webrtc\base\constructormagic.h(27) : see previous definition of 'DISALLOW_COPY_AND_ASSIGN'
FAILED: ninja -t msvc -e environment.x86 -- E:\b\build\goma/gomacc "E:\b\depot_tools\win_toolchain\vs2013_files\VC\bin\amd64_x86\cl.exe" /nologo /showIncludes /FC @obj\third_party\webrtc\modules\audio_processing\logging\audio_processing.aec_logging_file_handling.obj.rsp /c ..\..\third_party\webrtc\modules\audio_processing\logging\aec_logging_file_handling.cc /Foobj\third_party\webrtc\modules\audio_processing\logging\audio_processing.aec_logging_file_handling.obj /Fdobj\third_party\webrtc\modules\audio_processing.cc.pdb
e:\b\build\slave\win\build\src\base\macros.h(28) : error C2220: warning treated as error - no 'object' file generated
e:\b\build\slave\win\build\src\base\macros.h(28) : warning C4005: 'DISALLOW_COPY_AND_ASSIGN' : macro redefinition
e:\b\build\slave\win\build\src\third_party\webrtc\base\constructormagic.h(27) : see previous definition of 'DISALLOW_COPY_AND_ASSIGN'
FAILED: ninja -t msvc -e environment.x86 -- E:\b\build\goma/gomacc "E:\b\depot_tools\win_toolchain\vs2013_files\VC\bin\amd64_x86\cl.exe" /nologo /showIncludes /FC @obj\third_party\webrtc\modules\audio_processing\beamformer\audio_processing.nonlinear_beamformer.obj.rsp /c ..\..\third_party\webrtc\modules\audio_processing\beamformer\nonlinear_beamformer.cc /Foobj\third_party\webrtc\modules\audio_processing\beamformer\audio_processing.nonlinear_beamformer.obj /Fdobj\third_party\webrtc\modules\audio_processing.cc.pdb
e:\b\build\slave\win\build\src\base\macros.h(28) : error C2220: warning treated as error - no 'object' file generated
e:\b\build\slave\win\build\src\base\macros.h(28) : warning C4005: 'DISALLOW_COPY_AND_ASSIGN' : macro redefinition
e:\b\build\slave\win\build\src\third_party\webrtc\base\constructormagic.h(27) : see previous definition of 'DISALLOW_COPY_AND_ASSIGN'
Original issue's description:
> Consolidate constructormagic macros with Chromium version and remove Chromium override.
>
> Part of work removing dependency on Chromium's base.
>
> Only adds "= delete". From https://codereview.chromium.org/1151443003 :
> "This will guarantee the error to be at compile time, and not rely on the call visibility (private)."
>
> In consequence of that change, fixed an illegal copy and removed a bunch of unused variables.
>
> BUG=chromium:468375 (in particular comment #37)
> NOTRY=true
>
> Committed: https://crrev.com/0de8ff488d92e0bc6b7b65662898ff5e955cda93
> Cr-Commit-Position: refs/heads/master@{#9913}
TBR=andrew@webrtc.org,henrikg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:468375 (in particular comment #37)
Review URL: https://codereview.webrtc.org/1330283002
Cr-Commit-Position: refs/heads/master@{#9914}
/external/webrtc/webrtc/base/virtualsocketserver.cc
|
| 0de8ff488d92e0bc6b7b65662898ff5e955cda93 |
|
10-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Consolidate constructormagic macros with Chromium version and remove Chromium override.
Part of work removing dependency on Chromium's base.
Only adds "= delete". From https://codereview.chromium.org/1151443003 :
"This will guarantee the error to be at compile time, and not rely on the call visibility (private)."
In consequence of that change, fixed an illegal copy and removed a bunch of unused variables.
BUG=chromium:468375 (in particular comment #37)
NOTRY=true
Review URL: https://codereview.webrtc.org/1316363005
Cr-Commit-Position: refs/heads/master@{#9913}
/external/webrtc/webrtc/base/virtualsocketserver.cc
|
| 38f8893235f3b80ae9ca89db66d62ca819b51c01 |
|
14-Aug-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
WebRTC Bug 4865
Bug 4865: even without STUN/TURN, as long as the peer is on the open internet, the connectivity should work. This is actually a regression even for hangouts.
We need to issue the 0.0.0.0 candidate into Port::candidates_ and filter it out later. The reason is that when we create connection, we need a local candidate to match the remote candidate.
The same connection later will be updated with the prflx local candidate once the STUN ping response is received.
BUG=webrtc:4865
R=juberti@webrtc.org
Review URL: https://codereview.webrtc.org/1274013002 .
Cr-Commit-Position: refs/heads/master@{#9708}
/external/webrtc/webrtc/base/virtualsocketserver.cc
|
| be508a1d3634ce63b64cd740c44600453e3c3a6b |
|
06-Apr-2015 |
Guo-wei Shieh <guoweis@chromium.org> |
Implement Tcp Reconnect for TCPPort.
UDP case should not be changed.
Active TCPConnection will initiate Reconnect after OnClose and when Send or Ping fails.
Passive TCPConnection will prune itself as usual as the active side will create a new connection.
The Reconnect could make P2PCT choose a different best_connection in the case where connectivities exist b/w more than 1 Network.
Also, to avoid upper layer triggers ice restart, the WRITE_TIMEOUT caused by the socket disconnection is delayed to give the reconnect mechanism chance to kick in. The timeout event is only fired if the reconnect can't work in 5 sec. If the reconnect, there should be no ICE disconnected state trigger either in active or passive side.
BUG=1926
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31359004
Cr-Commit-Position: refs/heads/master@{#8929}
/external/webrtc/webrtc/base/virtualsocketserver.cc
|
| 67186fe00cc68cbe03aa66d17fb4962458ca96d2 |
|
09-Mar-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Fix clang style warnings in webrtc/base
Mostly this consists of marking functions with override when
applicable, and moving function bodies from .h to .cc files.
Not inlining virtual functions with simple bodies such as
{ return false; }
strikes me as probably losing more in readability than we gain in
binary size and compilation time, but I guess it's just like any other
case where enabling a generally good warning forces us to write
slightly worse code in a couple of places.
BUG=163
R=kjellander@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47429004
Cr-Commit-Position: refs/heads/master@{#8656}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8656 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/virtualsocketserver.cc
|
| d3b453be17d6f91c4e1f9a5544b7b2d52d448f81 |
|
14-Feb-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Remove the incremental IP address behavior from virtualsocketserver
VirtualSocketServer, when binding to any address (all 0s), will assign a unique IP address by incrementing the IP address, resulted in 0.0.0.1. However, this breaks the testing of 4276 where we bind to all 0s and expect the local address should remain all 0s.
BUG=4276
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35189004
Cr-Commit-Position: refs/heads/master@{#8370}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8370 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/virtualsocketserver.cc
|
| ff689be3c0c59c1be29aaa0697aa0f762566d6c6 |
|
12-Feb-2015 |
andresp@webrtc.org <andresp@webrtc.org> |
Use std::min and std::max instead of self-defined functions such as rtc::_min/_max.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35079004
Cr-Commit-Position: refs/heads/master@{#8347}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8347 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/virtualsocketserver.cc
|
| 53d9012faf32eb711681fdeb31b9d0d2f9e9481b |
|
09-Feb-2015 |
andresp@webrtc.org <andresp@webrtc.org> |
Clean kForever from basictypes and move it to the interfaces that actually have it.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33269004
Cr-Commit-Position: refs/heads/master@{#8296}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8296 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/virtualsocketserver.cc
|
| 95a32ec098b90695f309f5cbdb41f2eb489a9434 |
|
07-Feb-2015 |
bjornv@webrtc.org <bjornv@webrtc.org> |
Revert 8271 "VirtualSocketServer out-of-order issue with closing..."
Failed on Linux_Memcheck bot.
http://chromegw/i/client.webrtc/builders/Linux%20Memcheck/builds/3182
> VirtualSocketServer out-of-order issue with closing TCP sockets
>
> https://webrtc-codereview.appspot.com/41449004 added a TURN TCP
> allocation release test which was disabled as it triggered an assert
> in the turnserver.
>
> This was caused by VirtualSockerServer delivering the last TCP packet
> after closing the connection. Calling
> VirtualSocketServer::SendTcp
> and
> VirtualSocket::Close
> from TestTurnTCPReleaseAllocation led to the following order of
> messages in VirtualSocket::OnMessage:
> MSG_ID_DISCONNECT
> MSG_ID_PACKET
>
> This is out of order and triggers an assert in turnserver.cc since the
> socket from which the message arrives has already been discarded,
> subsequently breaking the test.
>
> In VirtualSocketServer::Disconnect the MSG_ID_DISCONNECT is posted to the
> msg_queue immediately, thus getting ahead of any (slightly delayed)
> actual packets.
>
> Maybe PostAt(network_delay_ + 1, ...) would be better?
>
> Re-enables TestTurnTCPReleaseAllocation.
>
> BUG=
> R=juberti@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/34759004
TBR=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38979004
Cr-Commit-Position: refs/heads/master@{#8280}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8280 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/virtualsocketserver.cc
|
| 4770437da965ce8ad85891e520a1275b9493c25d |
|
06-Feb-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
VirtualSocketServer out-of-order issue with closing TCP sockets
https://webrtc-codereview.appspot.com/41449004 added a TURN TCP
allocation release test which was disabled as it triggered an assert
in the turnserver.
This was caused by VirtualSockerServer delivering the last TCP packet
after closing the connection. Calling
VirtualSocketServer::SendTcp
and
VirtualSocket::Close
from TestTurnTCPReleaseAllocation led to the following order of
messages in VirtualSocket::OnMessage:
MSG_ID_DISCONNECT
MSG_ID_PACKET
This is out of order and triggers an assert in turnserver.cc since the
socket from which the message arrives has already been discarded,
subsequently breaking the test.
In VirtualSocketServer::Disconnect the MSG_ID_DISCONNECT is posted to the
msg_queue immediately, thus getting ahead of any (slightly delayed)
actual packets.
Maybe PostAt(network_delay_ + 1, ...) would be better?
Re-enables TestTurnTCPReleaseAllocation.
BUG=
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34759004
Cr-Commit-Position: refs/heads/master@{#8271}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8271 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/virtualsocketserver.cc
|
| 4fba293c87c366a3fd38ea94e88c3d38021f0dfa |
|
18-Dec-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Workaround for issue 3927 to allow localhost IP even if it doesn't match the local turn port
BUG=3927
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7941 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/virtualsocketserver.cc
|
| 0eb6eec5cb8f8157301d858908b6956a631f20be |
|
17-Dec-2014 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Move VirtualSocket into the .h file to allow unit tests more control over behavior.
BUG=3927
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7935 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/virtualsocketserver.cc
|
| 22406fcc9bd70de7dcf2a536ed464d458d940b63 |
|
09-Sep-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH.
BUG=3570
R=juberti@webrtc.org, mallinath@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7070
Review URL: https://webrtc-codereview.appspot.com/20999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7120 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/virtualsocketserver.cc
|
| f1427c673189971662d7cf2159195862640968f9 |
|
05-Sep-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Revert 7070 "TurnPort should retry allocation with a new address on error
STUN_ERROR_ALLOCATION_MISMATCH."
TBR=jiayl@webrtc.org
BUG=N/A
Review URL: https://webrtc-codereview.appspot.com/15359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7072 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/virtualsocketserver.cc
|
| 574f2f60feaa41f4ca5d36381129066e6e8c25cb |
|
04-Sep-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH.
BUG=3570
R=juberti@webrtc.org, mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7070 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/virtualsocketserver.cc
|
| f048872e915a3ee229044ec4bc541f6cbf9e4de1 |
|
13-May-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds a modified copy of talk/base to webrtc/base. It is the first step in
migrating talk/base to webrtc/base.
BUG=N/A
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17479005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6129 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/virtualsocketserver.cc
|
| e9a604accd54ab14dbf98f99ccdcf3ae1c54d27c |
|
13-May-2014 |
perkj@webrtc.org <perkj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 6107 "Adds a modified copy of talk/base to webrtc/base. I..."
This breaks Chromium FYI builds and prevent roll of webrtc/libjingle to Chrome.
http://chromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win%20Builder/builds/457
> Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base.
>
> BUG=N/A
> R=andrew@webrtc.org, wu@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/12199004
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6116 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/virtualsocketserver.cc
|
| 2c7d1b39b9374d2bc9bda4755fd4813db66a135c |
|
12-May-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base.
BUG=N/A
R=andrew@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6107 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/virtualsocketserver.cc
|