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Author SHA1 Message Date
pbos@webrtc.org
3d0019f09a Remove ViEBase::Init() call from VideoCall. 
ViEBase::Init() is a no-op in the current implementation. Keeping it
there is just confusing.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4544 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-14 14:27:11 +00:00
pbos@webrtc.org
fd39e13c80 Remove VideoEngine class from new VideoEngine API. 
The VideoEngine class had minimal use, so it makes more sense to bake
its functionality and config into VideoCall for a simpler API. The only
thing the VideoEngine class could do was to create VideoCalls.

BUG=2224
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2020004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4543 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-14 13:52:52 +00:00
pbos@webrtc.org
d65914360a Disable CanTransmitExtraRtpPacketsWithoutError on Windows. 
Flakily crashes on Windows.

BUG=2240
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2028005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4542 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-14 09:44:19 +00:00
marpan@webrtc.org
62ecc20afb Revert r4539 "Disable racy part of RunsRtpRtcpTestWithoutErrors". 
Bot failures for Win32-Release and Linux64-Release.

TBR=pbos@webrtc.org.

Review URL: https://webrtc-codereview.appspot.com/2026004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4541 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-13 21:36:48 +00:00
vikasmarwaha@webrtc.org
83ffb0dd5c Added functionality in apprtc demo to close the capture device on hangup. 
BUG=1589
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2018004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4540 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-13 17:53:37 +00:00
pbos@webrtc.org
a05653b2c1 Disable racy part of RunsRtpRtcpTestWithoutErrors. 
Disabled part as suggested in bug 1790, but without breaking it up into
multiple tests. These tests will be made redundant by tests for the new
API, and it would take far too long to clean these up properly.

BUG=1790
R=kjellander@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2022004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4539 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-13 14:27:20 +00:00
wuchengli@chromium.org
e1051b0731 Add native_handle.h to gyp. 
BUG=http://crbug.com/170345
TEST=Build all.
R=stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4538 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-13 05:53:38 +00:00
minyue@webrtc.org
db1cefc14e To allow the propagation of under-run in NetEq. 
BUG=
R=tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1974004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4537 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-13 01:39:21 +00:00
wu@webrtc.org
97d1a988b6 Remove suppressions for the cases that's already fixed. 
Rename some of the suppressions to new issue.
Fix leaks in virtualsocket_unittest.

BUG=1972,1976,2100
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2010005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4536 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-13 00:13:26 +00:00
wu@webrtc.org
6603736038 PeerConnection::RemoveStream now removes the local stream even when it's closed. Updated the unit test accordingly. 
RISK=P3
TESTED=PeerConnectionInterfaceTest.CloseAndTestMethods
TBR=fischman_webrtc

Review URL: https://webrtc-codereview.appspot.com/2018005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4535 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-13 00:09:35 +00:00
fischman@webrtc.org
32001ef124 PeerConnection shutdown-time fixes 
- TCPPort::~TCPPort() was leaking incoming_ sockets; now they are deleted.
- PeerConnection::RemoveStream() now removes streams even if the
  PeerConnection::IsClosed().  Previously such streams would never get removed.
- Gave MediaStreamTrackInterface a virtual destructor to ensure deletes on base
  pointers are dispatched virtually.
- VideoTrack.dispose() delegates to super.dispose() (instead of leaking)
- PeerConnection.dispose() now removes streams before disposing of them.
- MediaStream.dispose() now removes tracks before disposing of them.
- VideoCapturer.dispose() only unowned VideoCapturers (mirroring C++ API)
- AppRTCDemo.disconnectAndExit() now correctly .dispose()s its
  VideoSource and PeerConnectionFactory.
- CHECK that Release()d objects are deleted when expected to (i.e. no ref-cycles
  or missing .dispose() calls) in the Java API.
- Create & Return webrtc::Traces at factory birth/death to be able to assert
  that _all_ threads started during the test are collected by the end.
- Name threads attached to the JVM more informatively for debugging.
- Removed a bunch of unnecessary scoped_refptr instances in
  peerconnection_jni.cc whose only job was messing with refcounts.

RISK=P2
TESTED=AppRTCDemo can be ended and restarted just fine instead of crashing on camera unavailability.  No more post-app-exit logcat lines.  PCTest.java now asserts that all threads are collected before exit.

BUG=2183
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2005004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4534 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-12 23:26:21 +00:00
mallinath@webrtc.org
a5506690b4 Update libjingle to 50733053. 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2017004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4532 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-12 21:18:15 +00:00
pbos@webrtc.org
4ca7d3f9fe Replace MapWrapper with std::map<>. 
MapWrapper was needed on some platforms where STL wasn't supported, we
now use std::map<> directly.

BUG=2164
TEST=trybots
R=henrike@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2001004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4530 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-12 19:51:57 +00:00
fischman@webrtc.org
dd14b2add1 libjingle gyp: signal errors during gyp time to avoid cryptic failures during build time. 
- $JAVA_HOME / java_home missing or not pointing to a JDK
- Multiple or zero mac codesigning identities

BUG=2206
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2012004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4527 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-12 18:06:29 +00:00
elham@webrtc.org
1928d0ef67 Updated WebRTC version to 3.39 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2014004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4525 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-12 17:12:44 +00:00
pbos@webrtc.org
468e19aa93 Signal when shutting down DirectTransport. 
Avoids starting the network thread when there are no packets to be read.
This allows the transport to shut down directly, which makes tests using
it able to quit faster, and not have to wait up to 10ms.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2010004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4524 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-12 14:28:00 +00:00
wuchengli@chromium.org
0d94c2f81c Avoid acquiring VCM::_receiveCritSect during decode callback. 
When VideoDecoder::Decode, Reset, or Release is called,
VideoCodingModuleImpl::_receiveCritSect may have been
acquired. Decode callback needs to acquire the same lock
in ViEChannel::FrameToRender. It is not a problem for
SW decode because decode callback is run on the same
WebRTC decoding thread and the lock is re-entrant. But
for HW decode, decode callback is run on a thread different
from WebRTC decoding thread. Decode callback gets the locks
in the opposite order. Deadlock can happen.

BUG=http://crbug.com/170345
TEST=Try apprtc.appspot.com/?debug=loopback on ARM Chromebook Daisy.
     Run libjingle_peerconnection_unittest.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1997005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4523 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-12 14:20:49 +00:00
pbos@webrtc.org
9668467d87 Run loopback tests with network thread. 
Running with a network thread provides a more realistic simulation. Like
a real network, packets are handed off to a socket, or buffer, and then
the call returns. This prevents weird scenarios when both the sending
side and receiving side are on the call stack simultaneously, which can
cause deadlocks as locks could otherwise be taken simultaneously in both
the sender and receiver order by the same thread.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2000005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4522 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-12 12:59:04 +00:00
minyue@webrtc.org
ecbe0aa543 Added Opus stereo support 
TESTED=git try
BUG=webrtc:1360
R=tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1868004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4521 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-12 06:48:09 +00:00
wu@webrtc.org
91053e7c5a Update libjingle to 50654631. 
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2000006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4519 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-10 07:18:04 +00:00
sergeyu@chromium.org
bf853f2732 Fix crash in screen capturer on Mac 
BUG=crbug.com/247685
R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2006004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4518 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-10 01:30:23 +00:00
pbos@webrtc.org
6cd9341801 Hand over loopback packets to a network thread. 
This version of LoopBackTransport hands packets over to a network thread
which will deliver them instead. This allows SendRTP and SendRTCP to
always be able to return, preventing deadlocks in voe_auto_test. The
previous case did not represent actual network usage. Now the send and
receive side can run concurrently with the receiving side. Previously
the sender thread also drove the receiving side, which does not
represent the regular use case where packets are put on a network
socket.

BUG=1568,2081,2178
TEST=Ran VoiceEngine RtpRtcpTest.*, known for deadlocking, 100+ times.
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1985005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4516 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-09 21:11:57 +00:00
stefan@webrtc.org
80865fd611 Don't pace out packets or generate padding when the pacer is disabled. 
TEST=trybots
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2000004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4513 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-09 11:31:23 +00:00
pbos@webrtc.org
2ab209ef14 Remove include_dirs from test/test.gyp. 
This is a cleanup step for having root-relative includes, include_dirs shouldn't be needed anymore.

BUG=1662
R=phoglund@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1984004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4512 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-09 08:49:48 +00:00
pbos@webrtc.org
a3b7406219 Remove unused unreferenced code in webrtc/ 
The code removed here are .c, .cc and .h files that are not referenced
from anywhere else. E.g. if git-grep showed no occurrence of the file
it's removed. This process was repeated until no more unreferenced
files were present.

BUG=
R=andrew@webrtc.org, henrike@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org, turaj@webrtc.org, wu@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1945004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4511 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-09 08:47:51 +00:00
wuchengli@chromium.org
f4081ab8d8 Revert "Avoid acquiring VCM::_receiveCritSect during decode callback." 
This reverts commit aa3528a9cd65b176b9d6f9d58cecb1068891dca4.

BUG=http://crbug.com/170345
TEST=libjingle_peerconnection_unittest
TBR=stefan,wu

Review URL: https://webrtc-codereview.appspot.com/1999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4510 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-09 04:42:51 +00:00
wuchengli@chromium.org
a717ee9962 Avoid acquiring VCM::_receiveCritSect during decode callback. 
When VideoDecoder::Decode, Reset, or Release is called,
VideoCodingModuleImpl::_receiveCritSect may have been
acquired. Decode callback needs to acquire the same lock
in ViEChannel::FrameToRender. It is not a problem for
SW decode because decode callback is run on the same
WebRTC decoding thread and the lock is re-entrant. But
for HW decode, decode callback is run on a thread different
from WebRTC decoding thread. Decode callback gets the locks
in the opposite order. Deadlock can happen.

BUG=http://crbug.com/170345
TEST=Try apprtc.appspot.com/?debug=loopback on ARM Chromebook Daisy.
R=stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1977004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4509 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-09 04:08:38 +00:00
mikhal@webrtc.org
64799da6c6 Allowing decoding with errors, when disabling nack. 
BUG=1897
R=stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1982004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4508 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-08 22:45:33 +00:00
niklas.enbom@webrtc.org
e270331481 Fix duplicate code 
R=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1993004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4507 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-08 22:23:48 +00:00
mallinath@webrtc.org
5a27e49f35 This CL will add support of passing all turn urls returned by the CEOD to PeerConnection object. 
R=juberti@webrtc.org, vikasmarwaha@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4506 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-08 19:52:08 +00:00
pbos@webrtc.org
58d76cb635 Delete Channels without ChannelManager lock. 
Triggered Helgrind error, as deleting a Channel will also unregister a
module which has called GetChannel(), resulting in a cyclic lock graph.
This change will also allow other threads to access the ChannelManager
instance while Channels are deleted.

R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1946005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4505 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-08 17:32:21 +00:00
tina.legrand@webrtc.org
bd21fb5f8d Adding call to Opus PLC 
NetEq will call the PLC function in Opus only to set the decoder state. The actual PLC data will not be used.

BUG=https://code.google.com/p/webrtc/issues/detail?id=1181
R=tterribe@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1727004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4504 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-08 11:01:07 +00:00
agalusza@google.com
d177c10e2d Added logic for kSelectiveErrors to VCMJitterBuffer and corresponding unit tests. 
R=marpan@google.com, mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1943004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4503 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-08 01:12:33 +00:00
pbos@webrtc.org
676ff1ed89 Ref-counted rewrite of ChannelManager. 
The complexity of the last ChannelManager and potentially usage of it as well caused race conditions and deadlocks in loopback voe_auto_test. This ref-counted solution takes no long-term locks, uses less locks overall and is significantly easier to understand.

ScopedChannel has been split up into a ChannelOwner with a reference to a channel and an Iterator over ChannelManager. Previous code was really used for both things. ChannelOwner is used as a shared pointer to a channel object, while an Iterator should work as expected.

BUG=2081
R=tommi@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1802004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4502 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-07 17:57:36 +00:00
fischman@webrtc.org
825e9b0a9b talk/objc/README: s/libjingle/webrtc/ in repository path. 
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1985004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4501 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-07 16:52:03 +00:00
pbos@webrtc.org
a165d9c0a4 Code formatting on files touched in r4447. 
BUG=
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4500 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-07 14:17:05 +00:00
pwestin@webrtc.org
401ef361ac Added configuration of max delay to ACM and NetEq 
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1964004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4499 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-06 21:01:36 +00:00
fischman@webrtc.org
c883fdc273 PeerConnection.java: enable setting trace & log levels from Java 
Replaces the hard-coded scheme that was there before and lets apps decide what
to log and to where.

R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4498 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-06 19:00:53 +00:00
agalusza@google.com
c4e1ab515b Added Decoding with errors API to video_coding.h and removed unused DecodeError enum. 
R=marpan@google.com, mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1937004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4497 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-06 18:27:41 +00:00
turaj@webrtc.org
0fc2558503 Add turaj@webrtc.org to NetEq owners. 
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1980004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4496 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-06 17:07:18 +00:00
phoglund@webrtc.org
94aca5c7de Disabled flaky HardwareTest.BuiltInWasapiAECWorksForAudioWindowsCoreAudioLayer. 
TBR=xians@webrtc.org
BUG=2179

Review URL: https://webrtc-codereview.appspot.com/1955005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4495 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-06 08:20:47 +00:00
phoglund@webrtc.org
bd69d1beaf Disabled SsrcPropagatesCorrectly on Linux. 
BUG=2178
TBR=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1975004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4494 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-06 08:03:16 +00:00
minyue@webrtc.org
7bb5436e5d Better error treatment in NetEqImpl::InsertPacketInternal() 
BUG=webrtc:1364
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1844004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4493 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-06 05:40:57 +00:00
minyue@webrtc.org
9721db799c removed NetEq::EnableDtmf() 
BUG=webrtc:1373
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1822005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4492 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-06 05:36:26 +00:00
vikasmarwaha@webrtc.org
6e7c203aee Modified apprtc demo code to detect browser by checking user_agent in apprtc.py. Now we will use Mozilla stun server if FF is detected as the browser. The CL is an improvement to r4388. 
R=braveyao@webrtc.org, dutton@google.com

Review URL: https://webrtc-codereview.appspot.com/1928004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4491 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-05 22:05:20 +00:00
wu@webrtc.org
9dba525627 * Update libjingle to 50389769. 
* Together with "Add texture support for i420 video frame." from
wuchengli@chromium.org.
https://webrtc-codereview.appspot.com/1413004

RISK=P1
TESTED=try bots
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1967004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4489 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-05 20:36:57 +00:00
fischman@webrtc.org
f696f253b2 Invert dependency between webrtc_utility and media_file targets to reflect reality. 
BUG=2166
R=henrike@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1953004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4488 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-05 18:45:19 +00:00
elham@webrtc.org
9b8861c358 Updated WebRTC version number to 3.38 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1965004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4487 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-05 17:19:16 +00:00
pbos@webrtc.org
12dc1a38ca Switch C++-style C headers with their C equivalents. 
The C++ headers define the C functions within the std:: namespace, but
we mainly don't use the std:: namespace for C functions. Therefore we
should include the C headers.

BUG=1833
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1917004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4486 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-05 16:22:53 +00:00
fischman@webrtc.org
c3d93c6921 talk/PRESUBMIT: Accept copyright years going back to 2004. 
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1956004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4485 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-05 15:01:33 +00:00