kjellander@webrtc.org
6f8acdc76f
Suppressions for libjingle_unittest after roll in r5502
...
New errors arrived when rolling libjingle in r5502.
These suppressions are needed to green up the Memcheck and
TSan bots.
BUG=1976,2080
TEST=local runs on Linux:
tools/valgrind-webrtc/webrtc_tests.sh --tool=tsan -b out/Release -t libjingle_unittest
tools/valgrind-webrtc/webrtc_tests.sh --tool=memcheck -b out/Release -t libjingle_unittest
and trybot:
git try --bot=linux_memcheck,linux_tsan -t libjingle_unittest
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5509 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-08 21:15:59 +00:00
sergeyu@chromium.org
cc685acbdf
Disable AsyncInvokeTest.CancelInvoker test
...
Test is flaky.
BUG=b/12944358
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5508 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-08 19:59:00 +00:00
sergeyu@chromium.org
0178810659
Don't use LOG() in callback.h
...
Because chromium is compiled with a different version of logging macros
defined in logging.h that header cannot be used in headers that can
also included from chromium code. Removed LOG_F(LS_WARNING) from
callback.h . That issue would block this code from being rolled in
chromium.
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5507 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-08 03:18:03 +00:00
fischman@webrtc.org
bfc26dcc10
gitignore: ignore webrtc android demo apps build artifacts, and sort list
...
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8209004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5506 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-08 01:05:45 +00:00
mallinath@webrtc.org
5a59ccbb6d
Switching to NSS random number generator and adding init method to unittests.
...
R=jiayl@webrtc.org , sergeuy@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8259004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5505 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-07 23:22:00 +00:00
sergeyu@chromium.org
ad3035fc9e
Fix WindowCapturerWin to unselect bitmap before destroying DC.
...
BUG=https://code.google.com/p/webrtc/issues/detail?id=2901
R=wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/8229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5504 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-07 21:24:04 +00:00
vikasmarwaha@webrtc.org
c5a839c3a9
Updated demos so they work on FF, the createOffer api cannot have null parameters according to spec. Same applies to createAnswer.
...
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/8219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5503 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-07 19:08:38 +00:00
sergeyu@chromium.org
9cf037b831
Update libjingle to 61168196
...
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5502 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-07 19:03:26 +00:00
sprang@webrtc.org
9510e53cc0
Make VideoReceiveStream::GetStats() const.
...
BUG=
R=mflodman@webrtc.org , pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5501 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-07 15:32:45 +00:00
kjellander@webrtc.org
f7d4993940
Remove Coverity script.
...
Chromium has decided to drop Coverity so
we don't have any reason for maintaining this code.
Personally, I think that from a quality perspective other tools,
like all the new compiler warnings that are constantly being added
to the Clang compiler is a better way to address dangers in the code.
The maintenance cost and overhead of such advanced tools like Coverity
is simply too high.
BUG=
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7669007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5500 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-07 13:50:40 +00:00
sprang@webrtc.org
09315705b9
Wire up statistics in video receive stream of new API
...
This CL includes Call tests that test both send and receive sides.
BUG=2235
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5499 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-07 12:06:29 +00:00
kjellander@webrtc.org
90ce73a0d5
Remove svn:ignore for third_party/cygwin
...
In preparation for landing DEPS removal of Cygwin
in https://webrtc-codereview.appspot.com/8099004/
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5498 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-07 10:04:21 +00:00
vikasmarwaha@webrtc.org
b307e86076
Updated demos to use the sucess and failure callback in addIceCandidate api.
...
R=dutton@google.com
Review URL: https://webrtc-codereview.appspot.com/7969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5497 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-06 22:38:37 +00:00
fbarchard@google.com
60de116687
libyuv.gyp fix for ios sim which is intel not neon, fixing a link error.
...
BUG=none
TEST=try bots
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8069004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5496 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-06 21:17:16 +00:00
marpan@webrtc.org
dfef7ba971
Roll libvpx 241571:248011
...
TBR=ajm@google.com
Review URL: https://webrtc-codereview.appspot.com/8129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5495 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-06 18:40:35 +00:00
stefan@webrtc.org
77c917a6ee
Plot the capacity of a trace-based delivery filter.
...
Breaks out the instantaneous rate counters to its own class.
R=solenberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7999005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5494 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-06 16:34:47 +00:00
pbos@webrtc.org
ea1c5ad58f
Fix gunit compilation on VS2012.
...
In VS2012 compiling gunit or its dependencies triggers a lot of
"'std::tuple' : too many template arguments" warnings. The workaround
for this, done for gtest already, is to define _VARIADIC_MAX=10.
BUG=2616
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5493 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-06 13:17:20 +00:00
michaelbai@google.com
f928f5c87c
Use system's cpu_features library
...
Remove the copied cpu_featrues.c/h
Use the NDK's cpu_features.a or the one build from android source.
This issue blocked libvpx roll.
BUG=334447
R=andrew@webrtc.org , fischman@webrtc.org , henrike@webrtc.org , wjia@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8019004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5492 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-05 18:43:46 +00:00
stefan@webrtc.org
c88d3368d5
Add delay and send/receive throughput plots to BWE simulation.
...
R=mflodman@webrtc.org , solenberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5491 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-05 15:57:14 +00:00
henrik.lundin@webrtc.org
75642fcd9a
Implementing replacement audio support in neteq_rtpplay
...
This CL makes it possible to replace the payload in an RTP stream
with audio from another (PCM) file. The new payload will be encoded as
PCM16b. The RTP headers will be updated to reflect this change of
payload type.
BUG=2834
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5490 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-05 08:49:13 +00:00
henrik.lundin@webrtc.org
e6ab21b9ca
Fixing a bug in DummyRTPpacket
...
This bug caused writing outside allocated memory when RTP header
extensions were used.
BUG=2834
TBR=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8009005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5489 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-05 08:46:46 +00:00
andrew@webrtc.org
54744918ef
Update AudioProcessing::Create docs.
...
TBR=bjornv
Review URL: https://webrtc-codereview.appspot.com/8039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5488 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-05 06:30:29 +00:00
jiayl@webrtc.org
20a60ea39d
Fix a cursor capturing issue on Windows.
...
The input position to WindowFromPoint should be relative to the desktop, not
relative to the window; if the result from WindowFromPoint is a child window
of the shared top window, it should be captured.
BUG=
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/7959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5487 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-04 17:49:12 +00:00
stefan@webrtc.org
0e5a2b5de6
Handle the invalid case of setting multiple stream_bitrates if there is only a single send stream registered.
...
This can happen when switching between multiple streams and a single while getting feedback from the receiver.
BUG=2881
TEST=trybots
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5486 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-04 14:38:25 +00:00
pbos@webrtc.org
3e6c41c48f
Revert "Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEvents"
...
This reverts commit r5479.
R=henrika@webrtc.org
BUG=2880
Review URL: https://webrtc-codereview.appspot.com/7989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5485 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-04 10:45:14 +00:00
pbos@webrtc.org
064b32acbb
Fix locking in LoopBackTransport::StorePacket.
...
The critical section in StorePacket was unnamed and only existed in
expression scope. Added GUARDED_BY annotations (which caught the bug),
then fixed it by naming the variable.
BUG=2880
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5484 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-04 09:42:02 +00:00
andrew@webrtc.org
36291da197
Pull Chromium's clang-format binaries.
...
This gets 'git cl format' working again in a standalone webrtc checkout.
It started failing after this depot_tools change:
https://codereview.chromium.org/134313007
Depends on this change:
https://codereview.chromium.org/135653014/
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5483 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-04 01:45:10 +00:00
andrew@webrtc.org
f6a638e001
Trivial rename of non-compile time consts.
...
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7669006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5482 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-04 01:31:28 +00:00
marpan@webrtc.org
e88c186dbe
Revert r5480
...
TBR=ajm@google.com
Review URL: https://webrtc-codereview.appspot.com/7959005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5481 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-04 00:02:35 +00:00
marpan@webrtc.org
e35ecb476b
Roll libvpx 241571:248011
...
R=andrew@webrtc.org
TBR=ajm@google.com
Review URL: https://webrtc-codereview.appspot.com/7949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5480 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-03 22:53:10 +00:00
marpan@webrtc.org
f6b8f496ee
Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEvents
...
Issue: https://code.google.com/p/webrtc/issues/detail?id=2880
R=andrew@webrtc.org
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5479 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-03 21:34:35 +00:00
fischman@webrtc.org
6e08228525
PeerConnectionTest(java): remove the obsolete magical names of streams & tracks.
...
BUG=1253
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7929005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5478 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-03 19:15:44 +00:00
fischman@webrtc.org
a06ebab1e1
PeerConnectionTest(java): test SCTP DataChannels.
...
BUG=1408,2253,2626
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5477 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-03 19:11:29 +00:00
mallinath@webrtc.org
ecd622eec3
Updating libjingle.gyp after addition new files yuvframescapturer.cc.
...
TBR=pbos@webrc.org
Review URL: https://webrtc-codereview.appspot.com/7919006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5476 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-03 17:17:05 +00:00
mallinath@webrtc.org
67ee6b9a62
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
2014-02-03 16:57:16 +00:00
stefan@webrtc.org
422fdbf502
Wire up feedback to VideoSender.
...
BUG=
R=solenberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5474 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-03 16:33:50 +00:00
aluebs@webrtc.org
c9ee412070
Re-enabling audio processing tests
...
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5473 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-03 14:41:57 +00:00
xians@webrtc.org
c1e28038ba
Moved the new OnData interface to AudioTranport, and expose the AudioTransport pointer via voe_base
...
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5472 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-02 15:30:20 +00:00
jiayl@webrtc.org
1af5ea0538
Implement single monitor capture on Mac.
...
BUG=2787, 2824
TESTED=MacBook Pro Retina with an external monitor; verified changing display configuration while capturing; add/remove monitor while capturing; verified cursor position.
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/7479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5471 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-01 02:03:24 +00:00
henrik.lundin@webrtc.org
83aee8f450
Fixing test name for NetEqPerformanceTest
...
The naming did not follow conventions.
BUG=2859
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5469 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-31 11:46:34 +00:00
asapersson@webrtc.org
bdc5ed2e7d
Add configuration for cpu overuse detection to video send stream.
...
BUG=2422
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5468 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-31 10:05:07 +00:00
kjellander@webrtc.org
7d7f08957c
Add gyp_webrtc script to generate projects.
...
The reason for this is that http://crrev.com/245412
introduces a dependency of Chrome's src/build/gyp_chromium
to src/tools/find_depot_tools.py, which we don't have
synced in WebRTC (src/tools is very big).
Offline discussions shows that we cannot rely on syncing
individual subdirectories from Chrome in the future, but
maintaining our own gyp_webrtc file will at least buy us
some time for now, so we can roll past that chromium_revision
in WebRTC DEPS.
Overview of differences between gyp_webrtc and gyp_chromium
(and how we previously used gyp_chromium):
* No .gyp file needs to be passed (defaults to all.gyp)
* CHROMIUM_GYP_FILE is ignored (i.e. cannot be used to
specify an alternate .gyp file to process)
* Ninja is used by default on all platforms unless GYP_GENERATORS
is set.
* Gyp syntax check is always on
* Gyp circular dependency check is always on
* No support for automatic toolchain detection on Windows.
* --depth argument is no longer needed since calculated by
the script.
* Support for a webrtc.gyp_env file sitting next to the
.gclient file in the top dir of checkout, which can be
used to override Gyp variables similar to chromium.gyp_env.
* SKIP_WEBRTC_GYP_ENV can be set to skip reading webrtc.gyp_env.
BUG=2863
TEST=Ran and verified behavior on Linux with:
gclient runhooks
webrtc/build/gyp_webrtc
webrtc/build/gyp_webrtc -Dextra_gyp_flag=0
. build/android/envsetup.sh && gclient runhooks
SKIP_WEBRTC_GYP_ENV=1 webrtc/build/gyp_webrtc
GYP_GENERATORS=make webrtc/build/gyp_webrtc
The patch also passes runhooks and compile step on all trybots.
R=andrew@webrtc.org , fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7759004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5467 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-31 09:34:51 +00:00
stefan@webrtc.org
1dd9b4d98e
Add BWE tools for parsing RTP files.
...
bwe_rtp_play feeds packets from an RTP file into the BWE and prints the estimates.
bwe_rtp_to_text parses an RTP file and outputs the result to a text file.
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7689006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5466 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-31 09:15:48 +00:00
juberti@webrtc.org
668a23b402
Fix MIME type on new demo pages.
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5465 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-30 23:42:01 +00:00
juberti@webrtc.org
5db9a3f32a
Added new create-offer and ice-servers demos to test the exact output of createOffer and .onicecandidate.
...
Updated a few demos to work on Firefox.
R=dutton@google.com
Review URL: https://webrtc-codereview.appspot.com/1581006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5464 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-30 23:38:44 +00:00
jiayl@webrtc.org
bda5fa77af
Fix the mouse cursor offset issue on Mac.
...
The problem is that MouseCursorMonitor returns coordinates in DIPs, while DisplayAndMouseComposer assumes that they are in physical pixels. The fix is to convert the position to physical pixels in MouseCursorMonitorMac.
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/7739006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5463 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-30 23:27:35 +00:00
henrikg@webrtc.org
c693704cc2
Move out typing detection to its own class.
...
This will allow an embedder to use it directly.
Adding inertia/hangover time between updates of the reported detection status to the algorithm, controlled by a parameter. That is usually desired and this way a consumer of
the class don't have to implement that. (VoiceEngine will let it be 1, which results in the same behavior as before, and keep controlling the hangover itself.)
R=andrew@webrtc.org , niklas.enbom@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5462 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-30 09:50:46 +00:00
jiayl@webrtc.org
cf1b51b6fb
Moves the display reconfiguration callback into a separate class,
...
so that it can be shared with the cursor monitor when single monitor capturing
is added (https://webrtc-codereview.appspot.com/4679005/ ).
This Cl should have no functionality change.
BUG=2253
R=henrike@webrtc.org , sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/7599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5461 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-29 21:59:12 +00:00
jiayl@webrtc.org
808b99b111
Disable a test assert which fails due to usrsctp not cleaned up in SctpDataEngine.cc
...
BUG=2749
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7739005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5460 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-29 19:44:40 +00:00
jiayl@webrtc.org
a576faf82a
Enable SCTP and use OPENSSL on Anroid and NSS on other platforms.
...
It includes unit test fixes to properly initialize SSL if DTLS or SSL random number generator is used in the tests.
The private key and certificate constant strings used in some tests are updated to be compatible with NSS.
A few potentially overflow type conversions caused compiling warning on Windows and they are fixed by importing and using Chromium's checked_cast, which aborts the program if overflow occurs.
It also fixes a leak in nssstreamadapter.cc by releasing the PRFileDesc* in StreamClose.
BUG=2253
R=fischman@webrtc.org , juberti@google.com , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4679005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5459 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-29 17:45:53 +00:00