admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
7,075 Commits 1 Branch 0 Tags
Commit Graph

122 Commits

Author SHA1 Message Date
pthatcher@webrtc.org
5950b645b9 Use c++11 features in webrtc/base/network.cc as a test to see if we can use them. 
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25209005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7777 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-01 23:18:27 +00:00
pkasting@chromium.org
4591fbd09f Use size_t more consistently for packet/payload lengths. 
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.

This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.

BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom

Review URL: https://webrtc-codereview.appspot.com/23129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-20 22:28:14 +00:00
pbos@webrtc.org
f5b56fbc41 Annotate COMPILE_ASSERT with __attribute__((unused)). 
Also renames UNUSED -> ATTRIBUTE_UNUSED to be able to use this when
building peerconnection_jni.cc which apparently has this defined to
something else.

R=kjellander@webrtc.org
TBR=mflodman@webrtc.org
BUG=4018

Review URL: https://webrtc-codereview.appspot.com/28039005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7711 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-17 13:47:38 +00:00
jiayl@webrtc.org
0e71070207 Use ScreenCapturer to capture the whole and clip to the window rect when the shared window is on the top. 
BUG=crbug/403703, crbug/316603
R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/22759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7684 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-11 18:15:55 +00:00
henrike@webrtc.org
d89b69aade Fix WebRTC Win64 + BoringSSL build. 
There were many size_t to int conversions. RAND_poll and RAND_seed no longer do
anything in BoringSSL, so fix that one by removing it. Use a checked_cast for
the remaining ones.

BUG=chromium:429039
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7655 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 17:23:09 +00:00
henrike@webrtc.org
33a0e2d7ef Only configure the SSL library in one place. 
Build settings now use use_openssl in both Chromium and standalone builds. It
moves all the platform-specific SSL-related build checks to be conditioned on
this flag as appropriate.

This is to avoid colliding with Chromium's transition away from NSS.

This is a fixup of https://webrtc-codereview.appspot.com/29559004 to avoid
breaking use_legacy_ssl_defaults.

BUG=chromium:413497
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7537 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-27 18:13:40 +00:00
pbos@webrtc.org
e93cbd13d5 Fix data races in ThreadTest.ThreeThreadsInvoke. 
R=henrike@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/26819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7457 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 14:54:56 +00:00
henrike@webrtc.org
7f7b0a1cdd Re-enable ThreadCheckerDeathTest.MethodNotAllowedOnDifferentThreadInDebug (missed when enabling other base tests). 
BUG=3836
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/24909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7425 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-10 21:41:55 +00:00
xians@webrtc.org
e46bc77e94 Reland 28629004: adding new AEC dump start interface for chrome. 
This is required because we are not allow to pass CRT objects across dll boundaries, that says, when we pass a file descriptor from chrome dll to libpeerconnection dll, the file descriptor will become invalid immediate, more information can be found here:
http://msdn.microsoft.com/en-us/library/ms235460.aspx

R=andresp@webrtc.org, andrew@webrtc.org, bjornv@webrtc.org, henrike@webrtc.org, henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7418 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-10 08:36:56 +00:00
henrike@webrtc.org
c732a3e511 Re-enable allmost all base tests. 
BUG=3836
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/22989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7416 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-09 22:08:15 +00:00
henrike@webrtc.org
4a73519690 Re-enables a bunch of base unittests part II. 
BUG=3836
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/30709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7415 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-09 20:27:13 +00:00
henrike@webrtc.org
e30dab77df base/thread_unittest: wrap test was setting current thread to NULL. 
This broke unittests following ThreadTest.Wrap

BUG=3836
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7413 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-09 15:41:40 +00:00
pbos@webrtc.org
3ff788cf73 Increase timeout for AsyncWriteTest.TestWrite. 
Having a 10ms timeout for something meant to run on DrMemory is insane.

TBR=henrike@webrtc.org
BUG=3490

Review URL: https://webrtc-codereview.appspot.com/23959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7410 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-09 12:47:15 +00:00
pbos@webrtc.org
127ca3f8e5 Disable TestDTLSConnectWithSmallMtu on all platforms. 
Other bots elsewhere are breaking on this test, my money is on that this
might be due to different SSL versions being used on the different bots.
This test fails on at least a couple of bots that has use_openssl=1.

R=kjellander@webrtc.org
TBR=henrike@webrtc.org
BUG=3910

Review URL: https://webrtc-codereview.appspot.com/25839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7403 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-09 07:52:03 +00:00
henrike@webrtc.org
536eb98408 Re-enables a bunch of base unittests. 
BUG=3836
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7399 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-08 22:17:02 +00:00
kwiberg@webrtc.org
9c6dc46c6d CHECK/DCHECK: Explicitly state whether the condition can have side effects 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7394 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-08 12:19:56 +00:00
pbos@webrtc.org
27e5898f45 Explicitly unpoison FDs for MSan. 
MSan doesn't handle inline assembly that's used by FD_ZERO causing a
false positive.

R=earthdok@chromium.org, henrike@webrtc.org
BUG=chromium:344505

Review URL: https://webrtc-codereview.appspot.com/25799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7388 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-07 17:56:53 +00:00
pbos@webrtc.org
77d5a57e5c Revert "Only configure the SSL library in one place." 
This reverts commit r7378, which broke Windows compile targets
elsewhere.

R=kjellander@webrtc.org
TBR=henrike@webrtc.org
BUG=chromium:413497

Review URL: https://webrtc-codereview.appspot.com/28679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7384 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-07 11:43:03 +00:00
henrike@webrtc.org
8234fa6f0e Only configure the SSL library in one place. 
Build settings now use use_openssl in both Chromium and standalone builds. It
moves all the platform-specific SSL-related build checks to be conditioned on
this flag as appropriate.

This is to avoid colliding with Chromium's transition away from NSS.

BUG=chromium:413497
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7378 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-06 22:30:46 +00:00
jiayl@webrtc.org
7dfb7fa189 Reland disallowing blocking calls on the worker thread. 
This fixed the issue that invoking the call when the thread is not started.

BUG=3559
R=juberti@webrtc.org, thorcarpenter@google.com

Review URL: https://webrtc-codereview.appspot.com/24769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7325 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-29 22:45:55 +00:00
kjellander@webrtc.org
f21ea918ad GN: Add common configs to all targets. 
This is needed to ensure we have the same build with GN
as with GYP, since GYP includes the common.gypi on a global level.
Several fixes has been needed in the past because some code have
been built without the right defines.

BUG=3441
R=brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/28589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7317 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-28 17:37:22 +00:00
pbos@webrtc.org
34f2a9ea72 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
 2014-09-28 11:36:45 +00:00
pthatcher@webrtc.org
5d0071fb1f Build one of NSS or BoringSSL but not both. 
The libraries have some common symbols. When both are linked I observed NSS
SHA1_Update called followed by BoringSSL SHA1_Final, which results in a
segfault. We should only link one of these.

Based off of https://review.webrtc.org/25689004/

BUG=3855
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7310 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-26 18:53:40 +00:00
jiayl@webrtc.org
1fd362c31e Do not assert for blocking call allowed in Thread::Join. 
We do not allow blocking call from the worker thread, but on Android the worker thread may stop/join a SignalThread, which hits the assert.
AssertBlockingIsAllowedOnCurrentThread is used to make sure a thread does not do Invoke, so check that in Thread::Join does not seem to add much value.

BUG=3857
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7308 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-26 16:57:07 +00:00
jiayl@webrtc.org
f1d751c7de Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup. 
BUG=crbug/414211
R=juberti@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7293

Review URL: https://webrtc-codereview.appspot.com/22739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7301 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-25 16:38:46 +00:00
andresp@webrtc.org
37e1846d73 Revert "Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup." (rev 7293). 
Breaks windows bot as it was already showing on the try jobs on the

BUG=crbug/414211
R=jiayl@webrtc.org,juberti@webrtc.org
TBR=jiayl@webrtc.org,juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7294 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-25 07:30:14 +00:00
jiayl@webrtc.org
fe1eafb71a Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup. 
BUG=crbug/414211
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7293 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-24 21:13:39 +00:00
jiayl@webrtc.org
3987b6de50 Fix a problem in Thread::Send. 
Previously if thread A->Send is called on thread B, B->ReceiveSends will be called, which enables an arbitrary thread to invoke calls on B while B is wait for A->Send to return. This caused mutliple problems like issue 3559, 3579.
The fix is to limit B->ReceiveSends to only process requests from A.
Also disallow the worker thread invoking other threads.

BUG=3559
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7290 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-24 17:14:05 +00:00
pbos@webrtc.org
d60d79a145 Thread annotation of rtc::CriticalSection. 
Effectively re-lands r5516 which was reverted because talk/-only
checkouts existed. This now resides in webrtc/base/, so no talk/-only
checkouts should be possible.

This change also enables -Wthread-safety for talk/ and fixes a bug in
talk/media/webrtc/webrtcvideoengine2.cc where a guarded variable was
read without taking the corresponding lock.

R=andresp@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/27569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7284 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-24 07:10:57 +00:00
pbos@webrtc.org
38344ed280 Move thread_annotations.h to webrtc/base/. 
R=andresp@webrtc.org, mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/27579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7283 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-24 06:05:00 +00:00
tkchin@webrtc.org
c569a49a3d Unit tests for SSLAdapter 
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17309004

Patch from Manish Jethani <manish.jethani@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7269 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 05:56:44 +00:00
henrike@webrtc.org
88772874da Disabled several rtc_unittests so the tests can be turned on in the waterfall 
BUG=3836
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7254 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-22 07:30:48 +00:00
kjellander@webrtc.org
95705602bd Additional disabled tests in rtc_unittests. 
It appears https://review.webrtc.org/27559004/
not enough to get rtc_unittests up and running.
It's currently failing on Linux 32, Linux ASan
and Win SyzyASan bots.

BUG=3836
TBR=henrike@webrtc.org
TEST=Locally passing rtc_unittests on Linux Release
build with asan=1 and lsan=1 in GYP_DEFINES.

Review URL: https://webrtc-codereview.appspot.com/24659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7242 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 14:49:37 +00:00
kjellander@webrtc.org
34ac7762e0 Additional disabled tests in rtc_unittests. 
It appears https://review.webrtc.org/30449004 was
not enough to get rtc_unittests up and running.

BUG=3836
TEST=Locally passing rtc_unittests on Mac Debug.
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7241 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 13:47:47 +00:00
henrike@webrtc.org
fded02c164 base: disabled several base tests on Mac so that rtc_unittests can be turned back on 
BUG=N/A
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7240 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 13:10:10 +00:00
jiayl@webrtc.org
ba737cba1a Do not require synchronization access on the thread if called from rtc::Thread::WrapCurrent. 
The synchronization access is unnecessary for rtc::Thread::WrapCurrent (called from JingleThreadWrapper) since JingleThreadWrapper never calls rtc::Thread::Stop or rtc::Thread::Join. Failing to get the access caused crashes in Chrome since rtc::Thread::Current will be NULL when rtc::Thread::WrapCurrent fails.

rtc::ThreadManager::WrapCurrentThread still requires the synchronization access, since I am not sure if the callers (e.g. the plugin) depends on it.

BUG=crbug/413853
R=juberti@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7224 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-18 16:45:21 +00:00
guoweis@webrtc.org
369a637ac8 Implemented Network::GetBestIP() selection logic as following. 
1) return the first global temporary and non-deprecrated ones.
2) if #1 not available, return global one.
3) if #2 not available, use ULA ipv6 as last resort.

ULA stands for unique local address. They are only useful in a private
WebRTC deployment. More detail: http://en.wikipedia.org/wiki/Unique_local_address

BUG=3808

At this point, rule #3 actually won't happen at current
implementation. The reason being that ULA address starting with 0xfc 0r 0xfd will be grouped into its own Network. The result of that is WebRTC will have one extra Network to generate candidates but the lack of rule #3 shouldn't prevent turning on IPv6 since ULA should only be tried in a close deployment anyway.

R=jiayl@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7200

Committed: https://code.google.com/p/webrtc/source/detail?r=7201

Review URL: https://webrtc-codereview.appspot.com/31369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7216 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-17 22:37:29 +00:00
guoweis@webrtc.org
40c2aa36f2 Implemented Network::GetBestIP() selection logic as following. 
1) return the first global temporary and non-deprecrated ones.
2) if #1 not available, return global one.
3) if #2 not available, use ULA ipv6 as last resort.

ULA stands for unique local address. They are only useful in a private
WebRTC deployment. More detail: http://en.wikipedia.org/wiki/Unique_local_address

BUG=3808

At this point, rule #3 actually won't happen at current
implementation. The reason being that ULA address starting with 0xfc 0r 0xfd will be grouped into its own Network. The result of that is WebRTC will have one extra Network to generate candidates but the lack of rule #3 shouldn't prevent turning on IPv6 since ULA should only be tried in a close deployment anyway.

R=jiayl@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7200

Review URL: https://webrtc-codereview.appspot.com/31369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7201 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-16 20:29:41 +00:00
guoweis@webrtc.org
f8bff762d1 Implemented Network::GetBestIP() selection logic as following. 
1) return the first global temporary and non-deprecrated ones.
2) if #1 not available, return global one.
3) if #2 not available, use ULA ipv6 as last resort.

ULA stands for unique local address. They are only useful in a private
WebRTC deployment. More detail: http://en.wikipedia.org/wiki/Unique_local_address

BUG=3808

At this point, rule #3 actually won't happen at current
implementation. The reason being that ULA address starting with 0xfc 0r 0xfd will be grouped into its own Network. The result of that is WebRTC will have one extra Network to generate candidates but the lack of rule #3 shouldn't prevent turning on IPv6 since ULA should only be tried in a close deployment anyway.

R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7200 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-16 20:17:22 +00:00
kjellander@webrtc.org
44360200e3 Fix GN for rtc_base_approved target. 
In https://webrtc-codereview.appspot.com/22649004
a new target was introduced that duplicated some
source files, breaking the bots in
http://build.chromium.org/p/chromium.webrtc.fyi/waterfall

This updates the GN config to also remove them from
the target where they were moved from in base.gyp.

BUG=3806
TESTED=Trybots + Running GN in a Chromium checkout with
src/third_party/webrtc symlinked to the WebRTC checkout
with this CL applied + passing compile step.

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7192 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-16 11:16:12 +00:00
andrew@webrtc.org
6ae5a6d7fe Add a target for the approved subset of rtc_base. 
rtc_base drags in a bunch of unwieldly dependencies (e.g. nss and
json) not required for standalone webrtc (aka rtc/media). The root of
the problem appears to be that MessageQueue depends on a socket server.
(And since common.h -> logging.h -> thread.h -> messagequeue.h, this
dependency spreads quickly.)

This starts a new target for a "purified" subset of rtc_base. It adds
the files which are already being used, replacing the use of common.h
with checks.h. desktop_capture is a lost cause, and retains its
dependency on the full rtc_base.

The hope is that as additional components are desired they will be
cleaned and added to rtc_base_approved.

BUG=3806
R=andresp@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7188 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-16 01:03:29 +00:00
henrik.lundin@webrtc.org
18617cfde8 Fix ThreadChecker unittests when DCHECK_ALWAYS_ON is defined 
This requires two fixes:
1. Use DCHECK instead of assert in ThreadChecker's unittest.

2. Activate DCHECK when DCHECK_ALWAYS_ON in enabled.

Both these modifications are in line with Chromium's implementation.
The ThreadChecker unittest was changed to use assert instead of DCHECK
on the initial import (since WebRTC did not have a DCHECK back then).

BUG=3803
TEST=local out/{Debug,Release}/rtc_unittests built with and without DCHECK_ALWAYS_ON
R=andrew@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7178 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-15 11:19:35 +00:00
henrike@webrtc.org
c3c9015bc6 linux: remove stray libcrypto dependency 
Followup to CL 20049004, which looks like it added an unneeded -lcrypto
on linux.

BUG=3625
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7168 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-12 16:11:38 +00:00
henrike@webrtc.org
78b2d56ac6 Disable MethodNotAllowedOnDifferentThreadInDebug. 
BUG=3803
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7167 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-12 15:57:08 +00:00
andresp@webrtc.org
f7e5f22f98 Fix stack limit exceeded in http client. 
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7165 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-12 13:35:05 +00:00
kjellander@webrtc.org
665d861115 Restore webrtc_base target until r7140 is rolled into Chromium. 
In r7140 the webrtc_base target was renamed to rtc_base. This
breaks our FYI bots for rolling WebRTC in Chromium's DEPS.
By re-adding a None target named webrtc_base, this transition
should be smoother.

TBR=henrikg@webrtc.org,
TESTED=Passed build/gyp_chromium on a Chromium checkout with src/third_party/webrtc replaced by a mount like this:
cd /path/to/chromium/src
sudo mount --bind /path/to/webrtc/trunk/webrtc third_party/webrtc

Review URL: https://webrtc-codereview.appspot.com/23589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7150 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-11 09:22:13 +00:00
henrike@webrtc.org
1711104b8a Fix MSVC warnings about value truncations, webrtc/base/ edition. 
BUG=chromium:81439
TEST=none
R=henrike@webrtc.org, marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/20249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7143 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-10 22:10:24 +00:00
tpsiaki@google.com
67eabc0938 Add schannel webrtc_base build using a new use_schannel gyp variable. 
R=henrike@webrtc.org, thorcarpenter@google.com

Review URL: https://webrtc-codereview.appspot.com/28409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7141 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-10 18:06:47 +00:00
henrike@webrtc.org
b2efb6771c Put base tests in webrtc_tests.gyp 
BUG=N/A
R=andrew@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7140 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-10 17:28:19 +00:00
andrew@webrtc.org
4ca66d691e include cstdlib for free() and abort() 
This previous CL added uses of free() and abort() without including cstdlib:
https://webrtc-codereview.appspot.com/22449004

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23559004

Patch from Mostyn Bramley-Moore <mostynb@opera.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7127 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-10 03:24:36 +00:00