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Author SHA1 Message Date
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
pbos@webrtc.org
ccdcbae177 Fix implicit int->bool conversion in VideoSendStream::DeliverRtcp. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1963004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4484 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-05 13:25:51 +00:00
pbos@webrtc.org
4052370e89 Use RtpHeaderParser in VideoCall implementation. 
BUG=1827
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1962004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4483 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-05 12:49:22 +00:00
pbos@webrtc.org
bbb07e69e5 Glue code and tests for NACK in new VideoEngine API. 
The test works by randomly dropping small bursts of packets until enough
NACKs have been sent back by the receiver. Retransmitted packets are
never dropped in order to assure that all packets are eventually
delivered. When enough NACK packets have been received and all dropped
packets retransmitted, the test waits for the receiving side to send a
number of RTCP packets without NACK lists to assure that the receiving
side stops sending NACKs once packets have been retransmitted.

BUG=2043
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1934004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4482 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-05 12:01:36 +00:00
pbos@webrtc.org
7fb9ce0cf5 Fix send times in video_full_stack. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1947004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4481 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-05 09:29:50 +00:00
pbos@webrtc.org
735a7c8b93 Add back is.FrameProvider() call lost in r4194. 
BUG=2119
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1946004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4480 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-05 09:03:03 +00:00
wu@webrtc.org
94349552de Disable P2PTransportChannelTest.* on memcheck and tsan bots due to issue 1972. 
TBR=mallinath
BUG=1972
RISK=P3
TEST=with below cmd lines and disabled tests won't run
tools/valgrind-webrtc/webrtc_tests.sh --build_dir out/Debug --test libjingle_p2p_unittest --gtest_filter=P2PTransportChannelTest* --tool tsan
tools/valgrind-webrtc/webrtc_tests.sh --build_dir out/Debug --test libjingle_p2p_unittest --gtest_filter=P2PTransportChannelTest* --tool memcheck

Review URL: https://webrtc-codereview.appspot.com/1954004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4479 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-02 23:30:50 +00:00
andrew@webrtc.org
2cbb429323 Remove redundant conditions key. 
Gives an error when gyp is run with CHROMIUM_GYP_SYNTAX_CHECK=1.

TBR=henrike

Review URL: https://webrtc-codereview.appspot.com/1952004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4478 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-02 20:52:54 +00:00
turaj@webrtc.org
7df9706a01 Add one API for implementing Initial delay. 
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4475 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-02 18:07:13 +00:00
henrike@webrtc.org
89c674053e Adds all unittests to android NDK-APK framework. 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1872004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4474 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-02 16:53:47 +00:00
pbos@webrtc.org
51b2459d37 Add some virtual and OVERRIDEs in webrtc/common_audio/ 
BUG=163
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4473 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-02 11:44:38 +00:00
pbos@webrtc.org
9162080527 Fix some chromium-style warnings in webrtc/modules/audio_processing/ 
BUG=163
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1902004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4472 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-02 11:44:11 +00:00
wu@webrtc.org
4ebd8efc09 Supress libjingle_unittest fails on TSan. 
BUG=2080
TBR=mallinath

Review URL: https://webrtc-codereview.appspot.com/1943005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4471 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-02 00:14:41 +00:00
wu@webrtc.org
a054569c15 Fix memory leak in datachannel and its test. 
RISK=P3
TESTED=memcheck build
tools/valgrind-webrtc/webrtc_tests.sh --tool memcheck --test out/Debug/libjingle_peerconnection_unittest  --gtest_filter=SctpDataChannelTest*

R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1941005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4470 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-01 22:08:14 +00:00
wu@webrtc.org
0dc0f172a3 sscanf isn't safe with strings that aren't null-terminated. In such case, create a local copy that is null-terminated first. 
TESTED=GYP_DEFINES=build_for_tool=memcheck gclient runhooks
ninja -C out/Debug/ libjingle_unittest
tools/valgrind-webrtc/webrtc_tests.sh --tool memcheck --test out/Debug/libjingle_unittest  --gtest_filter=Http*

R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/1941004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4469 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-01 21:20:46 +00:00
sergeyu@chromium.org
17758e96c5 Fix crash in DesktopRegion::Intersect(). 
BUG=crbug.com/266933
R=alexeypa@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1938004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4468 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-01 19:51:04 +00:00
fischman@webrtc.org
86d7a198ec ObjC PeerConnection README: note workaround needed for crbug.com/248168 
BUG=2106
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1940004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4467 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-01 19:27:54 +00:00
fischman@webrtc.org
1bc1954174 AppRTCDemo: builds using ninja on iOS for simulator and device! 
Things included in this CL:
- updated READMEs to provide an exact/reproable set of steps for getting the app
  running.
- gyp changes to build the iOS AppRTCDemo sample app using gyp+ninja instead of
  the hand-crafted Xcode project (which has never worked in its checked-in
  form), including a gyp action to sign the sample app for deployment to an iOS
  device (the app can also be used in the simulator)
- deleted the busted hand-crafted Xcode project for the sample app
- updated the sample app to match the PeerConnection API that ended up landing
  (in a surprising twist of fate, the API landed quite a bit later than the
  sample app and this is the first time the CR-time changes in the API are
  reflected in the sample app)
- updated the sample app to reflect apprtc.appspot.com HTML/JS changes (equiv to
  the AppRTCClient.java changes in http://s10/47299162)
- picked up the iossim DEPS to enable launching the sample app in the simulator
  from the command-line.
- renamed some files to match capitalization of the classes they contain (Ice ->
  ICE) per ObjC naming guidelines.
- ran the files involved in this CL through clang-format to deal with xcode
  formatting craxy.

BUG=2106
RISK=P2
TESTED=unittest builds with ninja and passes on OS=mac; sample app builds with ninja and runs on simulator and device, though no audio flows from simulator/device (will fix in a follow-up CL)
R=andrew@webrtc.org, justincohen@google.com, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1874005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4466 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-01 18:29:45 +00:00
wu@webrtc.org
6abb750993 Delete gtest_exclude for asan which doesn't have effect with how the bots are setup now 
Add gtest_exclude for tsan to disable some flakey tests.
Change tsan suppression since the function name has been changed from DecodeWithErrors to DecodeErrorMode.

TBR=mallinath

Review URL: https://webrtc-codereview.appspot.com/1930004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4465 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-01 18:00:02 +00:00
pbos@webrtc.org
a2a2718a6c Fix some chromium-style warnings in webrtc/system_wrappers/ 
BUG=163
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1906004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4464 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-01 17:26:15 +00:00
agalusza@google.com
a7e360e89b Removed lines preventing simultaneous kHardNack and decoding with errors. Also made changes recommended by gcl lint (with the exception of changing non-const references to pointers). 
Propagated orthogonal API for decoding with errors from VideoCodingModule to VCMJitterBuffer.
Modified VCMJitterBuffer to allow three error modes: kNoErrors, kSelectiveErrors, kWithErrors.

R=marpan@google.com, mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1846004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4463 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-01 03:15:08 +00:00
wu@webrtc.org
d64719d895 Update libjingle to 50191337. 
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1885005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4461 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-01 00:00:07 +00:00
fischman@webrtc.org
d3ae3c7b1f Unbreak clang/android build of webrtc. 
TESTED=All target builds once more with clang=1.
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4460 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 23:53:07 +00:00
wu@webrtc.org
7fdbb1c832 We don't need to link with libssl.so when we already depend on openssl. 
This fixes the hidden-symbol linker warnings.

BUG=2149
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1927004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4459 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 22:41:36 +00:00
wu@webrtc.org
27c0408a16 Suppressing tsan errors on libjingle_unittest and libjingle_peerconnection_unittest. 
BUG=1205,2080
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1924004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4458 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 22:41:15 +00:00
fischman@webrtc.org
caa7024b86 PeerConnectionTest.java: build on android bots as well as linux ones. 
BUG=1796
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1921005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4455 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 21:56:30 +00:00
henrike@webrtc.org
a543114004 Removes no longer needed valgrind-libjingle folder. Was workaround for some bots using wrong valgrind script. 
TBR=wu@webrtc.org

BUG=2146

Review URL: https://webrtc-codereview.appspot.com/1920004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4454 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 17:53:39 +00:00
wu@webrtc.org
d40b4d9685 Fix libjingle memory bots by suppressing some of the errors. 
BUG=1205,2153
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1923004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4453 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 17:32:36 +00:00
mflodman@webrtc.org
d4412feeb0 Adding possibility to use encoding time when trigger underuse for frame based overuse detection. 
BUG=
TEST=Added unittest.
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1885004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4452 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 16:42:21 +00:00
xians@webrtc.org
09e8c47ee5 Merge r4374 from stable to trunk. 
r4374 was mistakenly committed to stable, so this is to re-merge back to trunk.

Store the sequence number in StopSend() and resume it in StartSend().

When restarting the microphone device, we call StopSend() first, then
StartSend() later. Since we reset sequence number in StopSend(), it sometimes
causes libSRTP to complain about packets being replayed. Libjingle work around
it by caching the sequence number in WebRtcVoiceEngine.cc, and call
SetInitSequenceNumber() to resume the sequence number before StartSend().Store the sequence number in StopSend() and resume it in StartSend().

When restarting the microphone device, we call StopSend() first, then
StartSend() later. Since we reset sequence number in StopSend(), it sometimes
causes libSRTP to complain about packets being replayed. Libjingle work around
it by caching the sequence number in WebRtcVoiceEngine.cc, and call
SetInitSequenceNumber() to resume the sequence number before StartSend().

This patch fixes this problem by storing the sequence number in StopSend(), and
resume it in StartSend(). So that we can remove the workaround in libjingle.

BUG=2102
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1922004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4451 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 16:30:19 +00:00