889 Commits

Author SHA1 Message Date
jbauch
250fc658c5 Lazily allocate output buffer for AsyncTCPSocket.
As a follow-up to https://codereview.webrtc.org/1737053006/ this CL further
improves memory usage by lazily allocating output buffers up to the passed
maximum size. This also changes the output buffer to a Buffer object.

BUG=

Review URL: https://codereview.webrtc.org/1741413002

Cr-Commit-Position: refs/heads/master@{#11801}
2016-02-28 23:06:47 +00:00
jbauch
3c1657658d Don't allocate buffers for listening sockets.
Listening sockets will not read/write directly, so they don't need buffers.

BUG=

Review URL: https://codereview.webrtc.org/1737053006

Cr-Commit-Position: refs/heads/master@{#11791}
2016-02-26 17:31:41 +00:00
kjellander
7324eb9e62 Revert of Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. (patchset #2 id:40001 of https://codereview.webrtc.org/1737593002/ )
Reason for revert:
Breaks GN in chromium.

Original issue's description:
> Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies.
>
> webrtc/audio/audio_sink.h is used by voice engine, but webrtc/audio is
> depending on voice engine, resulting in a cyclic dependency (which we
> don't detect since we have that check turned off, see webrtc:4243).
>
> BUG=webrtc:4243, webrtc:5589
> R=pbos@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org
> TBR=tommi@webrtc.org
>
> Committed: https://crrev.com/99b345c4e50c59a776c56949c17da3f50992f1a2
> Cr-Commit-Position: refs/heads/master@{#11766}

TBR=solenberg@webrtc.org,pbos@webrtc.org,perkj@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4243, webrtc:5589

Review URL: https://codereview.webrtc.org/1739783002

Cr-Commit-Position: refs/heads/master@{#11769}
2016-02-25 16:37:02 +00:00
jbauch
13041cf11f Add CopyOnWriteBuffer class
This CL introduces a new class CopyOnWriteBuffer that holds data in a
refcounted Buffer which is shared between copied CopyOnWriteBuffer to avoid
unnecessary allocations / memory copies.

BUG=webrtc:5155

Review URL: https://codereview.webrtc.org/1697743003

Cr-Commit-Position: refs/heads/master@{#11767}
2016-02-25 14:16:58 +00:00
kjellander@webrtc.org
99b345c4e5 Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies.
webrtc/audio/audio_sink.h is used by voice engine, but webrtc/audio is
depending on voice engine, resulting in a cyclic dependency (which we
don't detect since we have that check turned off, see webrtc:4243).

BUG=webrtc:4243, webrtc:5589
R=pbos@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org
TBR=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1737593002 .

Cr-Commit-Position: refs/heads/master@{#11766}
2016-02-25 14:12:48 +00:00
Peter Boström
a5d8e4eef5 Build SharedExclusiveLock in Chromium.
Partially un-breaks the Chromium FYI build.

TBR=jbauch@webrtc.org, tommi@webrtc.org
BUG=

Review URL: https://codereview.webrtc.org/1739713002 .

Cr-Commit-Position: refs/heads/master@{#11765}
2016-02-25 13:54:21 +00:00
kjellander@webrtc.org
a2644c06ee Disable tests failing under UBSan to enable deployment to main waterfall.
modules_unittests: https://build.chromium.org/p/client.webrtc.fyi/builders/Linux%20UBSan/builds/1138/steps/modules_unittests/logs/stdio
[ RUN      ] ByteIoTest.Test64SBitBigEndian
../../webrtc/modules/rtp_rtcp/source/byte_io_unittest.cc:34:33: runtime error: shift exponent 64 is too large for 64-bit type 'long'

rtc_unittests: https://build.chromium.org/p/client.webrtc.fyi/builders/Linux%20UBSan/builds/1138/steps/rtc_unittests/logs/stdio
[ RUN      ] IPAddressTest.TestCountIPMaskBits
../../webrtc/base/ipaddress.cc:415:20: runtime error: negation of -2147483648 cannot be represented in type 'int32_t' (aka 'int'); cast to an unsigned type to negate this value to itself

[ RUN      ] BandwidthSmootherTest.TestSampleRollover
../../webrtc/base/rollingaccumulator.h:73:22: runtime error: signed integer overflow: 2147483647 * 2147483647 cannot be represented in type 'int'

[ RUN      ] RandomNumberGeneratorTest.UniformSignedInterval
../../webrtc/base/random_unittest.cc:121:50: runtime error: signed integer overflow: 2147483647 - -2147483648 cannot be represented in type 'int'

rtc_media_unittests: https://build.chromium.org/p/client.webrtc.fyi/builders/Linux%20UBSan/builds/1138/steps/rtc_media_unittests/logs/stdio
[ RUN      ] VideoCommonTest.TestComputeScaleWithHighFps
../../webrtc/media/base/videocommon.cc:75:34: runtime error: signed integer overflow: 2621440 - -2147483648 cannot be represented in type 'int'

BUG=webrtc:5487, webrtc:5490, webrtc:5491
NOTRY=True
R=pbos@webrtc.org
TBR=henrika@webrtc.org

Review URL: https://codereview.webrtc.org/1727233005 .

Cr-Commit-Position: refs/heads/master@{#11764}
2016-02-25 13:23:29 +00:00
jbauch
9ccedc38f6 Reland: Prevent data race in MessageQueue.
The CL prevents a data race in MessageQueue where the variable "ss_" is
modified without a lock while sometimes read inside a lock.

Also thread annotations have been added to the MessageQueue class.

This was already reviewed and landed in https://codereview.webrtc.org/1675923002/
but failed in Chromium GN builds due to sharedexclusivelock.cc not being
compiled in these builds. This changed in https://codereview.webrtc.org/1712773003/
so the reland should work fine now.

BUG=webrtc:5496

Review URL: https://codereview.webrtc.org/1729893002

Cr-Commit-Position: refs/heads/master@{#11758}
2016-02-25 09:15:05 +00:00
Jon Hjelle
6140fcc11c Move RTCFileLogger to webrtc/base/objc.
BUG=
R=jiayl@webrtc.org, tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1692243003 .

Patch from Jon Hjelle <hjon@andyet.net>.

Cr-Commit-Position: refs/heads/master@{#11754}
2016-02-25 00:33:22 +00:00
guoweis
4cc9f98e4c Fix bug 574524: DtlsTransportChannel crashes after SSL closes remotely
When remote side closes, opensslstreamadapter could return SR_EOS which will not trigger upper layer to clean up what's left in the StreamInterfaceChannel. The result of this is when there are more packets coming in, the Write on the StreamInterfaceChannel will overflow the buffer.

The fix here is that when receiving the remote side close signal, we also close the underneath StreamInterfaceChannel which will clean up the queue to prevent overflow.

BUG=574524
TBR=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1566023002

Cr-Commit-Position: refs/heads/master@{#11751}
2016-02-24 19:10:09 +00:00
ossu
b01c7816a8 Added functional variants of Buffer::SetData and Buffer::AppendData.
They are invoked with the maximum size of the data to be added, and a
callable that generates that data, like this:

buffer.AppendData(10, [] (rtc::ArrayView<uint8_t> av) {
    for (uint8_t i = 0; i != 5; ++i)
      av[i] = i;

    return 5;
  });

The callable returns the number of bytes actually written, and the
final Buffer size will be adjusted accordingly. SetData and AppendData
both return the number of bytes added (i.e. the return value of the
callable).

These versions will be useful when converting AudioEncoder::Encode to use Buffer rather than raw pointers.

Also added a few tests for the new functionality.

Review URL: https://codereview.webrtc.org/1717273002

Cr-Commit-Position: refs/heads/master@{#11733}
2016-02-24 09:06:02 +00:00
tkchin
f75d008235 Bitrate controller for VideoToolbox encoder.
Also fixes a crash on encoder Release.

BUG=webrtc:4081

Review URL: https://codereview.webrtc.org/1660963002

Cr-Commit-Position: refs/heads/master@{#11729}
2016-02-24 06:49:48 +00:00
Peter Boström
7ddc9deb4d Reduce the scope of rtc::Event::Wait() locking.
Reduces contention on event_mutex_ while taking gettimeofday(). Impact
highly hypothetical at this point, but less locking is better.

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1716563003 .

Cr-Commit-Position: refs/heads/master@{#11706}
2016-02-22 10:32:02 +00:00
jbauch
a18f638ab1 Include "sharedexclusivelock.cc" in Chromium GN build.
Landing https://codereview.webrtc.org/1675923002/ broke some Chromium FYI bots
because the GN build didn't include "sharedexclusivelock.cc" in that scenario.

This CL moves the files from the non-Chromium block into the common sources
list.

BUG=webrtc:5496

Review URL: https://codereview.webrtc.org/1712773003

Cr-Commit-Position: refs/heads/master@{#11699}
2016-02-21 09:56:23 +00:00
jbauch
9674d7cb89 Revert of Prevent data race in MessageQueue. (patchset #3 id:40001 of https://codereview.webrtc.org/1675923002/ )
Reason for revert:
Broke chromium.webrtc.fyi bots:
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/9891
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20GN/builds/11416

Fails with
-----
Undefined symbols for architecture x86_64:
  "rtc::SharedExclusiveLock::LockShared()", referenced from:
      rtc::MessageQueue::DoDestroy() in librtc_base.a(messagequeue.o)
      rtc::MessageQueue::socketserver() in librtc_base.a(messagequeue.o)
      rtc::MessageQueue::WakeUpSocketServer() in librtc_base.a(messagequeue.o)
      rtc::MessageQueue::Quit() in librtc_base.a(messagequeue.o)
      rtc::MessageQueue::Get(rtc::Message*, int, bool) in librtc_base.a(messagequeue.o)
      rtc::MessageQueue::Post(rtc::MessageHandler*, unsigned int, rtc::MessageData*, bool) in librtc_base.a(messagequeue.o)
      rtc::MessageQueue::DoDelayPost(int, unsigned int, rtc::MessageHandler*, unsigned int, rtc::MessageData*) in librtc_base.a(messagequeue.o)
      ...
  "rtc::SharedExclusiveLock::UnlockShared()", referenced from:
      rtc::MessageQueue::DoDestroy() in librtc_base.a(messagequeue.o)
      rtc::MessageQueue::socketserver() in librtc_base.a(messagequeue.o)
      rtc::MessageQueue::WakeUpSocketServer() in librtc_base.a(messagequeue.o)
      rtc::MessageQueue::Quit() in librtc_base.a(messagequeue.o)
      rtc::MessageQueue::Get(rtc::Message*, int, bool) in librtc_base.a(messagequeue.o)
      rtc::MessageQueue::Post(rtc::MessageHandler*, unsigned int, rtc::MessageData*, bool) in librtc_base.a(messagequeue.o)
      rtc::MessageQueue::DoDelayPost(int, unsigned int, rtc::MessageHandler*, unsigned int, rtc::MessageData*) in librtc_base.a(messagequeue.o)
      ...
  "rtc::SharedExclusiveLock::SharedExclusiveLock()", referenced from:
      rtc::MessageQueue::MessageQueue(rtc::SocketServer*, bool) in librtc_base.a(messagequeue.o)
ld: symbol(s) not found for architecture x86_64
-----

Looks like these are compiling without "webrtc/base/sharedexclusivelock.cc".

Original issue's description:
> Prevent data race in MessageQueue.
>
> The CL prevents a data race in MessageQueue where the variable "ss_" is
> modified without a lock while sometimes read inside a lock.
>
> Also thread annotations have been added to the MessageQueue class.
>
> BUG=webrtc:5496
>
> Committed: https://crrev.com/df88460372e7ce78c871a87774d7e6d82aac6ee3
> Cr-Commit-Position: refs/heads/master@{#11683}

TBR=ivoc@webrtc.org,pthatcher@webrtc.org,deadbeef@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5496

Review URL: https://codereview.webrtc.org/1714463003

Cr-Commit-Position: refs/heads/master@{#11686}
2016-02-19 15:16:19 +00:00
jbauch
df88460372 Prevent data race in MessageQueue.
The CL prevents a data race in MessageQueue where the variable "ss_" is
modified without a lock while sometimes read inside a lock.

Also thread annotations have been added to the MessageQueue class.

BUG=webrtc:5496

Review URL: https://codereview.webrtc.org/1675923002

Cr-Commit-Position: refs/heads/master@{#11683}
2016-02-19 15:03:36 +00:00
ossu
728012e49f Changed the semantics of Buffer::Clear to not alter the capacity
Also added a test for Clear to ensure this invariant holds.

With this change, it is easy to empty a Buffer and reuse its storage. Further down the line, code filling data into a Buffer could be written to just append to it, with the caller determining if the Buffer should first be cleared or not.

There is currently only one use of Buffer::Clear (in AudioEncoderCopyRed::Reset()) and it should benefit from the change, by not requiring a reallocation after Reset.

Review URL: https://codereview.webrtc.org/1707693002

Cr-Commit-Position: refs/heads/master@{#11680}
2016-02-19 10:38:37 +00:00
tkchin
ee75c7a78f Compile rtc_base_objc for Mac.
BUG=

Review URL: https://codereview.webrtc.org/1705513002

Cr-Commit-Position: refs/heads/master@{#11661}
2016-02-17 22:45:00 +00:00
honghaiz
e3c6c82717 When doing continual gathering, remove the local ports when a corresponding network is dropped.
BUG=

Review URL: https://codereview.webrtc.org/1696933003

Cr-Commit-Position: refs/heads/master@{#11660}
2016-02-17 21:00:35 +00:00
kwiberg
b7f89d6e66 Replace scoped_ptr with unique_ptr in webrtc/voice_engine/
Also introduce a pair of scoped_ptr <-> unique_ptr conversion
functions. By using them judiciously, we can keep these CL:s small and
avoid having to convert enormous amounts of code at once.

BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1702983002

Cr-Commit-Position: refs/heads/master@{#11658}
2016-02-17 18:04:26 +00:00
Torbjorn Granlund
a3dc79e072 Move SSLIdentity Generate() implementations from .h to .cc file.
This amends https://codereview.webrtc.org/1683193003/

BUG=
R=hbos@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1701953002 .

Cr-Commit-Position: refs/heads/master@{#11632}
2016-02-16 12:34:04 +00:00
torbjorng
e8dc081c35 Implement certificate lifetime parameter as required by WebRTC RFC.
BUG=chromium:569005

Review URL: https://codereview.webrtc.org/1683193003

Cr-Commit-Position: refs/heads/master@{#11629}
2016-02-15 17:36:01 +00:00
tommi
04af839a88 Move refcount.h and scoped_ref_ptr.h to rtc_base_approved.
BUG=

Review URL: https://codereview.webrtc.org/1701533002

Cr-Commit-Position: refs/heads/master@{#11615}
2016-02-14 16:11:17 +00:00
kwiberg
8fb3557052 rtc::Buffer: Replace an internal rtc::scoped_ptr with std::unique_ptr
We'd like to completely replace rtc::scoped_ptr with std::unique_ptr.
This is a first trial CL to see if using unique_ptr causes any
problems.

(As a side effect of removing the scoped_ptr.h include in buffer.h,
I had to fix broken includes in no less than three files.)

BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1687833006

Cr-Commit-Position: refs/heads/master@{#11588}
2016-02-11 21:36:57 +00:00
jbauch
541f1869ca Cleanup temporary files created by tests.
This CL removes some temporary files created by OptionsFileTest and
TransientFileUtilsTest.

BUG=

Review URL: https://codereview.webrtc.org/1688553002

Cr-Commit-Position: refs/heads/master@{#11554}
2016-02-10 17:10:00 +00:00
jbauch
097d54956d Added thread annotations to FifoBuffer.
This CL adds thread annotations to FifoBuffer and adds a missing CritScope
for attribute access that is modified in locked code paths.

Review URL: https://codereview.webrtc.org/1677333002

Cr-Commit-Position: refs/heads/master@{#11535}
2016-02-09 10:30:43 +00:00
jbauch
25d1f28fa9 Fix race between Thread ctor/dtor and MessageQueueManager registrations.
This CL fixes a race where for Thread objects the parent MessageQueue
constructor registers the object in the MessageQueueManager even though
the Thread is not constructed completely yet. Same happens during
destruction.

BUG=webrtc:1225

Review URL: https://codereview.webrtc.org/1666863002

Cr-Commit-Position: refs/heads/master@{#11497}
2016-02-05 08:25:04 +00:00
kjellander
988d31eb9b Move gtest_prod_util.h out of webrtc/test tree.
This is needed because the target is defined in webrtc/common.gyp
and its current location crosses package boundaries when generating
projects for some build systems.

NOTRY=True

Review URL: https://codereview.webrtc.org/1665603003

Cr-Commit-Position: refs/heads/master@{#11496}
2016-02-05 08:23:57 +00:00
jbauch
f2a2bf4ae4 Stay writable after partial socket writes.
This CL fixes an issue where the "writable" flag didn't stay set after
::send or ::sendto only sent a partial buffer.

Also SocketTest::TcpInternal has been updated to use rtc::Buffer instead
of manually allocating data.

BUG=webrtc:4898

Review URL: https://codereview.webrtc.org/1616153007

Cr-Commit-Position: refs/heads/master@{#11480}
2016-02-04 00:45:38 +00:00
tkchin
d1fb26d457 Add iOS tracing.
BUG=

Review URL: https://codereview.webrtc.org/1650993004

Cr-Commit-Position: refs/heads/master@{#11469}
2016-02-03 09:51:22 +00:00
honghaiz
a7ad7c3ca0 Get the adapter type information from Android OS.
BUG=

Review URL: https://codereview.webrtc.org/1594673002

Cr-Commit-Position: refs/heads/master@{#11463}
2016-02-02 20:54:28 +00:00
tommi
ae695e95a6 Refactor RtpSender and SSRCDatabase.
* SSRCDatabase doesn't need to be a global instance, so I've changed it to be a "regular" class (i.e. construct via ctor, not maybe via GetSSRCDatabase( + release via ReturnSSRCDatabase())).  If we ever have parallel tests running in the same process, they won't have the problem of using the same ssrc database.

* Made RtpSender a more const.  Also added some todos for myself and holmer to look into clarifying the threading model.

* Switched from CriticalSectionWrapper to rtc::CriticalSection

* Changed the random seeding to use TickTime::Now().Ticks() since TimeInMicroseconds() could return 0 when the process was starting.  This is what TimeInMicroseconds() does anyway but now we don't need to access a global clock object.

BUG=webrtc:3062

Review URL: https://codereview.webrtc.org/1623543002

Cr-Commit-Position: refs/heads/master@{#11462}
2016-02-02 16:34:16 +00:00
asapersson
799379e8c2 Let a minimum time interval pass (one bucket size) after initialization before reporting rates (to avoid rates being based on too short time intervals).
BUG=chromium:570038

Review URL: https://codereview.webrtc.org/1582333008

Cr-Commit-Position: refs/heads/master@{#11455}
2016-02-02 09:47:05 +00:00
conceptgenesis
3f70562bbb Fix WebRtc ninja x86 build using Visual Studio 2015 (set GYP_MSVS_VERSION=2015).
Visual Studio 2015 balks at the implicit truncation of values. Easily fixed with an explicit cast.

Fixed redefinition of CLOCKS_PER_SEC when using Visual Studio 2015 and the Windows 10 SDK. CLOCKS_PER_SEC is also defined in "<WIN10 SDK DIR>\include\10.0.10240.0\ucrt\time.h" and also has the value of 1000

Hiding snprintf definition if building with Visual Studio 2015

Fixed C4573 compiler complaint in audio_processing_impl_locking_unittest.cc.

BUG=webrtc:5183

Review URL: https://codereview.webrtc.org/1412653006

Cr-Commit-Position: refs/heads/master@{#11434}
2016-01-30 22:40:52 +00:00
Erik Språng
1c3909899d Use rtc::time for all your timing needs!
Initial step of unifying so that base/timeutils.h and Clock/TimeTime
from system_wrappers use the same implementation.

BUG=webrtc:5463
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1639543005 .

Cr-Commit-Position: refs/heads/master@{#11394}
2016-01-27 11:55:44 +00:00
Peter Boström
0b518bf6fc Remove incorrect cast to AsyncSocketAdapter.
socket_ in OpenSSLAdapter should be (and is in tests) an AsyncSocket but
doesn't have to be an AsyncSocketAdapter. In tests this is
rtc::VirtualSocket which is an rtc::AsyncSocket. This also matches the
BIO_new_socket type signature.

This fixes the remaining UBSan vptr bot errors.

BUG=webrtc:5124, webrtc:5226
R=tommi@webrtc.org, torbjorng@webrtc.org

Review URL: https://codereview.webrtc.org/1639883002 .

Cr-Commit-Position: refs/heads/master@{#11391}
2016-01-27 11:35:52 +00:00
Sergey Ulanov
fab0a2886d Fix BasicNetworkManager not to spam logs when internet is unreacheable.
BasicNetworkManager attemps to connect an UDP socket and logs an error
when connect() fails, e.g. when internet is not connected. These
errors are not very useful in the logs, but apper there every time
it attemps to refresh network list. Replaced the log statement with
LOG(LS_INFO).

R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1635823004 .

Cr-Commit-Position: refs/heads/master@{#11389}
2016-01-27 06:13:04 +00:00
sprang
e791ffd638 Remove non-monotonic clock support
Real time clock may cause problems as they can move (even backwards) if
the clock is changed, eg updated by NTP.

Non-monotonic clocks still in use on some platform (I'm looking at you,
Apple) for timed waits, but that should be less of an issue than actual
timestamps.

BUG=webrtc:5452

Review URL: https://codereview.webrtc.org/1613013002

Cr-Commit-Position: refs/heads/master@{#11375}
2016-01-26 09:53:24 +00:00
pbos
5ad935cb56 Remove mutable from rtc::CriticalSection members.
rtc::CriticalSection is now lockable from const methods and no longer
need to remain mutable.

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1613643004

Cr-Commit-Position: refs/heads/master@{#11367}
2016-01-25 11:52:53 +00:00
tommi
7d0d0e0763 Remove dead code from webrtc/base/timing.*
BUG=

Review URL: https://codereview.webrtc.org/1626253002

Cr-Commit-Position: refs/heads/master@{#11366}
2016-01-25 11:09:32 +00:00
tommi
7406b96abc CriticalSection: Use types+methods from base/platform_thread*.*.
Use PlatformThreadRef, CurrentThreadRef and IsThreadRefEqual instead of pthread_t, pthread_self and operator== (or !=).

BUG=

Review URL: https://codereview.webrtc.org/1619153003

Cr-Commit-Position: refs/heads/master@{#11355}
2016-01-22 13:13:38 +00:00
tommi
ed281e9c9b New lock implementation for mac.
According to my measurements, it's about 100x faster than the native mutex implementation in OSX.  Google "OSX mutex performance" for more info.

BUG=

Review URL: https://codereview.webrtc.org/1594723003

Cr-Commit-Position: refs/heads/master@{#11352}
2016-01-22 07:47:30 +00:00
Peter Boström
af9e6637c0 Make rtc::CriticalSection lockable from f() const.
Removes the use of mutable rtc::CriticalSection across the code.

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1611223002 .

Cr-Commit-Position: refs/heads/master@{#11342}
2016-01-21 15:57:03 +00:00
Jon Hjelle
da99da81c9 Update API for Objective-C RTCPeerConnectionFactory.
BUG=
R=jiayl@webrtc.org, tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1558473002 .

Patch from Jon Hjelle <hjon@andyet.net>.

Cr-Commit-Position: refs/heads/master@{#11326}
2016-01-20 21:40:35 +00:00
kjellander
3c85cad1d4 Roll chromium_revision 7a4fb8d..f527e86 (370025:370073)
Change log: 7a4fb8d..f527e86
Full diff: 7a4fb8d..f527e86

No dependencies changed.

Clang was updated 255169:257953.
Details: 7a4fb8d..f527e86/tools/clang/scripts/update.py

NOTRY=True
NOPRESUBMIT=True

Review URL: https://codereview.webrtc.org/1593713013

Cr-Commit-Position: refs/heads/master@{#11301}
2016-01-19 12:47:24 +00:00
honghaiz
cec0a08275 Add a new interface for creating a udp socket in which it binds the socket to a network if the network handle is set.
Plus, in stunport, turnport and allocation sequence, create a socket using the new interface.

BUG=

Review URL: https://codereview.webrtc.org/1556743002

Cr-Commit-Position: refs/heads/master@{#11279}
2016-01-15 22:49:15 +00:00
guoweis
56271ed889 fix bug 5430
Fixed misusage of Connection function and also fixed the test case.

BUG=webrtc:5430

Review URL: https://codereview.webrtc.org/1592763003

Cr-Commit-Position: refs/heads/master@{#11278}
2016-01-15 22:45:11 +00:00
torbjorng
79a5a83e10 Adapt to boringssl's new defaults.
This is now a merge with patchset #2 of https://codereview.webrtc.org/1550773002 after that CL was reverted.

BUG=webrtc:5381

Review URL: https://codereview.webrtc.org/1589493004

Cr-Commit-Position: refs/heads/master@{#11273}
2016-01-15 15:16:54 +00:00
kjellander
fcfc804e43 Eliminate defines in talk/
Replace LINUX, OSX and IOS defines with WEBRTC_ prefixed versions.
Remove no longer used defines from talk/build/common.gypi due to
previously migrated sources (into webrtc/p2p and webrtc/libjingle).

When this is rolled into Chromium, we can also clean up the platform
defines in
https://code.google.com/p/chromium/codesearch#chromium/src/third_party/libjingle/libjingle.gyp

NOTRY=True
BUG=webrtc:5420
TESTED=Ran all compile trybots with --clobber flag.
TBR=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1588453005

Cr-Commit-Position: refs/heads/master@{#11254}
2016-01-14 19:01:25 +00:00
sprang
3542013f58 Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ )
Reason for revert:
We're getting boringssl version conflicts. Reverting for now.

Original issue's description:
> Update with new default boringssl no-aes cipher suites. Re-enable tests.
>
> This undoes https://codereview.webrtc.org/1533253002 (except the DEPS part).
>
> BUG=webrtc:5381
> R=davidben@webrtc.org, henrika@webrtc.org
>
> Committed: https://crrev.com/31c8d2eac5aec977f584ab0ae5a1d457d674f101
> Cr-Commit-Position: refs/heads/master@{#11250}

TBR=davidben@webrtc.org,henrika@webrtc.org,torbjorng@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5381

Review URL: https://codereview.webrtc.org/1586183002

Cr-Commit-Position: refs/heads/master@{#11253}
2016-01-14 17:14:06 +00:00