generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity
5,005 Commits 1 Branch 0 Tags 108 MiB
11aab0edc2
Commit Graph

5005 Commits

This Branch
This Branch
All Branches
Author SHA1 Message Date
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
xians@webrtc.org
07e5196414 Added new capture callback interface to pass the capture callback to a specific voe channel from libjingle webrtcvoiceengine.cc. 
The callback has to go through VoEBaseImpl since VoEChannel is internal to voice engine.

TEST=compile
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7769005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5458 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-29 13:54:02 +00:00
solenberg@webrtc.org
094ac39b5a Fix race when deleting video receive streams in Call. 
BUG=
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5457 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-29 11:21:58 +00:00