Like PlatformThreadId, this type is borrowed from Chromium.
The difference between the two is that PlatformThreadRef is pthread_t on posix platforms.
On Windows PlatformThreadRef and PlatformThreadId are the same thing.
The reason for this switch is pretty crazy. On Chromium's "Mac 10.9 dbg" bot,
we have been seeing the following code:
ThreadCheckerImpl::ThreadCheckerImpl() : valid_thread_(CurrentThreadId()) {
fprintf(stderr, "*** valid=%d\n", valid_thread_);
valid_thread_ = CurrentThreadId();
fprintf(stderr, "*** valid after=%d\n", valid_thread_);
}
print this:
*** valid=946872320
*** valid after=5647
This is for the same thread checker instance.
What's worse is that printing out what CurrentThreadId was returning, yielded that it was always returning 5647.
After switching over to pthread_t on Mac, this stopped happening.
So, to remove the current hack, reinstate the class on Mac and take a look at the next problem, I'm switching to pthread_t.
Really looking forward to truly getting to the bottom of this.
Tbr-ing since the build is essentially broken (we can't roll).
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37199004
Cr-Commit-Position: refs/heads/master@{#8283}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8283 4adac7df-926f-26a2-2b94-8c16560cd09d
I'm reverting the patch due to compilation issues. It would be great if we could make sure Chromium is ready for the patch before we land it in WebRTC.
As is, we're trying to roll webrtc into Chromium and we can't (this is not the only reason though). I might reland this after the roll, depending on how that goes though.
Here's an example failure:
e:\b\build\slave\win_gn\build\src\jingle\glue\channel_socket_adapter_unittest.cc(77) : error C2259: 'jingle_glue::MockTransportChannel' : cannot instantiate abstract class
due to following members:
'bool cricket::TransportChannel::GetSslCipher(std::string *)' : is abstract
e:\b\build\slave\win_gn\build\src\third_party\webrtc\p2p\base\transportchannel.h(107) : see declaration of 'cricket::TransportChannel::GetSslCipher'
ninja: build stopped: subcommand failed.
> This CL adds an API to the SSL stream adapters and transport channels to get the SSL cipher that was negotiated with the remote peer.
>
> BUG=3976
> R=davidben@chromium.org, juberti@webrtc.org, pthatcher@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/26009004TBR=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40689004
Cr-Commit-Position: refs/heads/master@{#8282}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8282 4adac7df-926f-26a2-2b94-8c16560cd09d
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/34759004TBR=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
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
In order to figure out the issue with the Mac 10.9 debug bot, this patch disables the ThreadChecker class on Mac in debug builds. For diagnostic purposes, it instead prints out when there's a thread mismatch. I'm also adding a DCHECK in case fetching the current thread id ever returns 0.
R=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40679004
Cr-Commit-Position: refs/heads/master@{#8269}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8269 4adac7df-926f-26a2-2b94-8c16560cd09d
This is the same change as already made for Windows:
https://webrtc-codereview.appspot.com/37069004/
* Remove "dead" and "alive" variables.
* Remove critical section
* Remove implementation of SetNotAlive()
* Always set thread name
* Add thread checks for correct usage.
* Changed AudioDeviceMac to create/start/stop/delete thread objects for playout and recording, inside the respective start and stop method. The reason for this is because the AudioDeviceMac instance is currently being created on one thread and the above Start/Stop methods are being called on a different thread. So, my change makes creation, start/stop, deletion of the thread objects always happen on the same thread.
I'm making CurrentThreadId() in rtc_base_approved more visible so that it can be used from there instead of inside webrtc. Down the line we will have more thread concepts in rtc_base_approved, so I put a TODO for myself to move this functionality to there once we do.
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40599004
Cr-Commit-Position: refs/heads/master@{#8235}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8235 4adac7df-926f-26a2-2b94-8c16560cd09d
The latter file was more up-to-date. The files are now identical
with the following exceptions:
* The namespace used (rtc vs. webrtc).
* The name of the include guard.
* base/scoped_ptr.h still has two extra methods, accept() and use().
* base/scoped_ptr.h still includes webrtc/base/common.h even though
it doesn't need it itself, since several .cc files expect to get
it for free by incuding base/scoped_ptr.h. This is of course bad
manners, and the "unused" include will be removed in a future CL.
A later CL will remove system_wrappers/interface/scoped_ptr.h.
R=andrew@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=8147
And reverted again, because out-of-tree code using this file was defining nullptr to 0: https://code.google.com/p/webrtc/source/detail?r=8149
Review URL: https://webrtc-codereview.appspot.com/36919004
Cr-Commit-Position: refs/heads/master@{#8196}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8196 4adac7df-926f-26a2-2b94-8c16560cd09d
Currently, in ipc_network_manager.cc, the UMA WebRTC.PeerConnection.IPv4Interfaces and its IPv6
counter part counts the addresses, instead of the interfaces as when
chromium delivers available networks to WebRTC, each address is wrapped
inside an individual network object.
The plan is to replace the current MergeNetworkList with the new one once it's rolled into chromium.
BUG=
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36779004
Cr-Commit-Position: refs/heads/master@{#8188}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8188 4adac7df-926f-26a2-2b94-8c16560cd09d
To do this, I'm removing ThreadChecker's dependency on the 'Thread' class, so that the checker works with any thread and doesn't rely on TLS.
Also simplifying CriticalSection's implementation on Windows since a critical section on Windows already knows what thread currently owns the lock.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8151 4adac7df-926f-26a2-2b94-8c16560cd09d
The latter file was more up-to-date. The files are now identical
with the following exceptions:
* The namespace used (rtc vs. webrtc).
* The name of the include guard.
* base/scoped_ptr.h still has two extra methods, accept() and use().
* base/scoped_ptr.h still includes webrtc/base/common.h even though
it doesn't need it itself, since several .cc files expect to get
it for free by incuding base/scoped_ptr.h. This is of course bad
manners, and the "unused" include will be removed in a future CL.
A later CL will remove system_wrappers/interface/scoped_ptr.h.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8147 4adac7df-926f-26a2-2b94-8c16560cd09d
This patch basically deletes webrtc/base/template_util.h (which is the
more outdated copy, although there are only cosmetical differences)
and moves webrtc/system_wrappers/source/template_util.h to take its
place.
The reunified header uses the rtc namespace like the old
webrtc/base/template_util.h, rather than the webrtc namespace like
webrtc/system_wrappers/source/template_util.h.
R=aluebs@webrtc.org, andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8050 4adac7df-926f-26a2-2b94-8c16560cd09d
1) Added SetMode() to SSLAdapter and OpenSSLAdapter so the mode can be set to
SSL_MODE_DTLS
2) OpenSSLAdapter overrides SendTo() and RecvFrom() to handle calls from
TurnPort via AsyncUdpSocket
3) OpenSSLAdapter derives from MessageHandler to implement an internal DTLS
timer
4) Updated SSLAdapter unit tests
BUG=
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19059004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7981 4adac7df-926f-26a2-2b94-8c16560cd09d
