919f87fb36
Delete capturers after destroying streams in test.
...
Since the renderers in CallTest.SendsAndReceiveStreams also stopped the
capturers they must be deleted after the VideoReceiveStream is stopped
or an use-after-free may occur.
R=mflodman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/5969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5300 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 14:55:54 +00:00
e7b1e11283
Revert 5285 "Revert 5228 "Use the RTT from RtcpRttStats class if..."
...
> Revert 5228 "Use the RTT from RtcpRttStats class if provided whe..."
>
> > Use the RTT from RtcpRttStats class if provided when sending/receiving NACK.
> >
> > R=holmer@google.com
> >
> > Review URL: https://webrtc-codereview.appspot.com/5049004
>
> TBR=asapersson@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/5799004
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5299 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 14:40:36 +00:00
1e7d61270c
Simplification of histogram normalization in delay estimator.
...
- Replaces a for loop with a single element update to save complexity. No regression in performance seen on set of recordings.
- Removes UpdatesMadeUponChange() and put code straight into ProcessBinarySpectrum().
BUG=None
TESTED=module_unittest, trybots, verified manually on set of recordings.
R=aluebs@webrtc.org , andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5298 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 13:37:28 +00:00
5ab756703e
Revert r5294 to re-roll r5293.
...
To fix races in test each stream now owns its own encoder/decoder.
R=mflodman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/5919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5297 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 12:24:44 +00:00
5c64508b03
Adds robust validation functionality to the delay estimator
...
Evaluated over a 51 recordings:
False positives went from 4.4% to 0.7%
Missed detections unchanged at 0.8%
No increase in complexity, but need to re-evaluate that.
TESTED=trybots, unittests, verified against Matlab implementation
BUG=None
R=aluebs@webrtc.org , andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5296 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 10:57:53 +00:00
87ad57bc75
Incorrect iterator++ in ModuleRtpRtcpImpl::RegisterVideoBitrateObserver
...
The iterator is incremented both in loop header and loop body. Should
only be incremented in header.
BUG=2727
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5295 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 07:43:51 +00:00
41e2615e02
Revert 5293 "Auto instantiate RBE depending on whether AST or TO..."
...
> Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream.
>
> BUG=
> R=mflodman@webrtc.org , stefan@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/5409004
TBR=solenberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5294 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-15 18:42:32 +00:00
341e91441a
Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream.
...
BUG=
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5293 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 23:57:54 +00:00
e1bc6c8d8b
Disabled tests on Android. The issue 2723 is filed to investigate the reason for tests failing.
...
BUG=
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5809005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5292 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 22:04:18 +00:00
dd393e7b9d
Measure pacer queue size based on when packets are inserted rather than captured.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5291 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 22:03:27 +00:00
167b6dfc73
Fix jitter buffer delay estimate.
...
BUG=b/12099925
R=niklas.enbom@webrtc.org , niklase@google.com , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5289 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 21:05:07 +00:00
24301a67c6
Update talk to 58174641 together with http://review.webrtc.org/4319005/ .
...
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5287 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 19:17:43 +00:00
92c2793154
Adding REMB to receive stream configuration, the send side will always
...
react to incoming REMB for now.
Adding a test to verify the receive side is generating RTCP REMB and
will follow up with a send side test as soon as the bitrate stats are
wired up for the new API.
TEST=See above.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5286 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 16:36:28 +00:00
86bb56a7f5
Revert 5228 "Use the RTT from RtcpRttStats class if provided whe..."
...
> Use the RTT from RtcpRttStats class if provided when sending/receiving NACK.
>
> R=holmer@google.com
>
> Review URL: https://webrtc-codereview.appspot.com/5049004
TBR=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5285 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 16:16:45 +00:00
0a222eba69
Merge metrics_unittests into video_engine_tests.
...
metrics_unittests will be removed as soon as trybots catch up with LKGR,
that way we don't have to break any tryjobs during.
BUG=1843
R=kjellander@webrtc.org , mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5284 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 14:31:47 +00:00
1d096901ac
Move realtime tests to webrtc_perf_tests.
...
New binary not to be run on our VMs as they result in flaky tests. These
will instead be run on baremetal machines.
BUG=2710
R=kjellander@webrtc.org , mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5283 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 12:48:05 +00:00
62451dcba0
Update talk to 58157731.
...
R=wu@webrtc.org
TBR=wu@webrc.org
Review URL: https://webrtc-codereview.appspot.com/5339005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5282 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 12:29:34 +00:00
6811b6e308
Callback for send bitrate estimates - new roll
...
Issue https://webrtc-codereview.appspot.com/4459004/ was commited as
r5259, after which flakiness was detected and a rollback was performed
at r5261.
Patch Set 1 of this issue is the code submitted in r5259. Subsequent
patch sets fixes a race condition which caused the seen problems.
The root cause was a dead lock between a thread sending rtp packets and
and a timed module processing thread:
webrtc::RTPSender::BitrateUpdated() // Get RTPSender stats lock
webrtc::Bitrate::Process() // Get Bitrate lock
webrtc::RTPSender::ProcessBitrate()
webrtc::ModuleRtpRtcpImpl::Process()
...
webrtc::Bitrate::Update() // Get Bitrate lock
webrtc::RTPSender::UpdateRtpStats() // Get RTPSender stats lock
webrtc::RTPSender::SendToNetwork()
...
This is fixed in Bitrate::Process() by releasing the lock before
calling the callback.
BUG=2235
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5281 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 09:46:59 +00:00
f3973e81d5
Make sure channels in the same call are in the same channel group.
...
Tested manually. I'll make a follow CL with a proper test once review.webrtc.org/5619004 has been committed.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5280 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 09:40:45 +00:00
e9abd591d7
Making RemoteRateControl::min_configured_bit_rate_ configurable
...
The minimum bitrate can now be configured from WrappingBitrateEstimator.
BUG=2698
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5279 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 08:42:42 +00:00
a9890800e0
Update talk to 58127566 together with
...
https://webrtc-codereview.appspot.com/5309005/ .
R=mallinath@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5277 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 00:21:03 +00:00
a92baead39
ACM 2 compatibility with ACM 1.
...
Removing an unregisterd codec from ACM 1 does not result in an error, so should be for ACM 2. Also ACM 1 has post-decode VAD on and AMC 2 needs to have it on by default.
BUG=
Test=trybits
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5549005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5276 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 00:10:44 +00:00
2018269dc3
Revert 5274 "Update talk to 58113193 together with https://webrt ..."
...
> Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/ .
>
> R=mallinath@webrtc.org , niklas.enbom@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/5719004
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5275 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-12 22:54:25 +00:00
a129b6cd13
Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/ .
...
R=mallinath@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5274 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-12 22:40:39 +00:00
451745ec05
Complete rewrite of demo application.
...
BUG=2122
R=andrew@webrtc.org , fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5273 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-12 16:55:37 +00:00
88ac63abc6
Remove overloaded CpuOveruseMeasure function.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5199005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5272 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-12 14:37:33 +00:00
df7b1d6e39
AppRTCDemo(android): make ant be quiet on success and not overly noisy on failure.
...
Also silence a 'cd' that would otherwise emit the path/to/talk.
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5271 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 22:36:22 +00:00
9ee75e9c77
Enables mixing and matching Java and native audio. It is used for getting best of both worlds capabilities (AEC and low latency).
...
BUG=N/A
R=fischman@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5270 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 21:42:44 +00:00
f41f06b916
PeerConnection(java): rationalize pointer-to-jlong conversion.
...
In r4665 I went a bit crazy with the manual reinterpretation of a pointer to a
jlong (to avoid undefined behavior) but that's what reinterpret_cast<> is for.
So use it directly now.
Added a do-nothing DataChannel to AppRTCDemo to regression test this, since the
only repro I've found of the original bug requires ARM ABI (PeerConnectionTest
on ia32 fails to repro).
BUG=2302
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5269 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 21:07:18 +00:00
9caf2765b2
Update talk to 58037405.
...
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/5579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5267 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 18:25:07 +00:00
391b4db7de
Fix common_video_unittests in apk_tests.gyp.
...
r5265 moved common_video_unittests to its own gyp, this required an
update of apk_tests.gyp that wasn't caught by our trybots.
TBR=mflodman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/5629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5266 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 16:48:53 +00:00
724947b8ef
Add SwapFrame() to VideoSendStreamInput.
...
Optionally prevents doing a frame copy when putting frames into a
VideoSendStream. PutFrame() is still there, which copies the frame.
Also removes time_since_capture_ms as a parameter, since
I420VideoFrame::render_time_ms() denotes when the frame was captured.
BUG=2657
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5265 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 16:26:16 +00:00
4c3faa9d73
Disable a libjingle unittest which is failing after a chromium roll out.
...
TBR=kjellander@google.com
BUG=
Review URL: https://webrtc-codereview.appspot.com/5559007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5264 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 16:12:31 +00:00
df02283279
Adds audio volume demo to the index page.
...
BUG=
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5589005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5263 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 14:44:10 +00:00
59d5705385
Fix memory tools error introduced in roll @ r5260
...
Turns out that the Chromium revision
https://src.chromium.org/viewvc/chrome?view=rev&revision=237238
introduced a new flag for the memory wrapper scripts.
Due to the way we reuse the chrome_tests.py for WebRTC purposes,
we need to add that flag too.
TEST=linux_tsan bot and locally running:
tools/valgrind-webrtc/webrtc_tests.sh --test test_support_unittests --tool tsan --target Release --build-dir out
from trunk/
BUG=none
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5589006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5262 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 14:16:53 +00:00
096e8d9f94
Revert 5259 "Callback for send bitrate estimates"
...
CL is causing flakiness in RampUpTest.WithoutPacing.
> Callback for send bitrate estimates
>
> BUG=2235
> R=mflodman@webrtc.org , pbos@webrtc.org , stefan@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/4459004
R=mflodman@webrtc.org , pbos@webrtc.org
TBR=mflodman
Review URL: https://webrtc-codereview.appspot.com/5579005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5261 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 14:07:33 +00:00
f9bdbe3619
Roll chromium_revision 232627:238260
...
This brings us the updated swarming_client
that has moved out from Chromium into a standalone
project.
Because of this, all .isolate files needed to be
updated as well, similar to the changes in
https://codereview.chromium.org/29993003
TEST=trybots passing
BUG=none
R=andrew@webrtc.org , perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5260 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 13:37:12 +00:00
2656cf9f4c
Callback for send bitrate estimates
...
BUG=2235
R=mflodman@webrtc.org , pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5259 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 12:53:03 +00:00
26c40ba166
Removed audio element from volume measuring demo.
...
This removes the possibility of feedback loops, which can happen if you
run this demo on an Android device.
BUG=
R=dutton@google.com
Review URL: https://webrtc-codereview.appspot.com/5589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5258 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 11:12:39 +00:00
1133ffda4b
Merged OWNERS of JS demo directories
...
This allows Sam Dutton to maintain code samples, and demo managers to
modify js/base/adapter.js.
BUG=
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5549006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5257 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 08:51:56 +00:00
c4038d795d
Rewriting the SoundMeter class to be RMS and be encapsulated differently
...
This CL changes the SoundMeter to be root-mean-square.
It also changes the interface between the meter and the display to be based on the display calling down to the meter rather than the meter calling up to the display.
A graphic display of the results is also added.
BUG=
R=cwilso@google.com , dutton@google.com , henrika@webrtc.org , juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5256 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 08:36:16 +00:00
77507eff4f
Correctly define OVERRIDE when building with g++ 4.7 and C++11 support
...
g++ 4.7 and later support explicit virtual overrides when building with C++11 support
enabled. However, libjingle does not detect that and makes OVERRIDE a no-op.
This CL updates base/common.h to define OVERRIDE properly when g++ 4.7 is used with
C++11 support enabled.
See this page for GCC support of C++11 features:
http://gcc.gnu.org/projects/cxx0x.html
R=fischman@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5159004
Patch from Chris Dumez <ch.dumez@samsung.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5255 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 00:07:11 +00:00
7ae8495779
Removed unnecessary Pulse init from VoE startup.
...
Saves 10% (~260ms) of the total PeerConnectionTest wallclock time.
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5254 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-10 21:01:34 +00:00
762fcdcca9
Correctly define OVERRIDE when building with g++ 4.7 and C++11 support
...
g++ 4.7 and later support explicit virtual overrides when building with C++11
support enabled. However, webrtc does not detect that and makes OVERRIDE a
no-op.
This CL updates typedefs.h to define OVERRIDE properly when g++ 4.7 is used
with C++11 support enabled.
See this page for GCC support of C++11 features:
http://gcc.gnu.org/projects/cxx0x.html
R=andrew@webrtc.org , fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5149005
Patch from Chris Dumez <ch.dumez@samsung.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5253 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-10 19:20:46 +00:00
8b8819262f
Improve VideoSendStreamTest::MaxPacketSize
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This CL was submitted as issue https://webrtc-codereview.appspot.com/4849004/ , but was reverted because of flakiness. This new issue will correct that.
Patch Set 1 contains the code that was submitted in 4849004.
BUG=2428
R=pbos@webrtc.org , phoglund@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5251 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-10 10:05:17 +00:00
917306d3fd
Change uses of the obsolete armv7 setting to arm_version==7.
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BUG=http://crbug.com/234135
R=andrew@webrtc.org , fischman@webrtc.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5369004
Patch from Mostyn Bramley-Moore <mostynb@opera.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5250 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-10 09:26:07 +00:00
eb7def234e
Fix compilation errors on Fedora 20.
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peerconnection_jni.cc: syscall() comes from <unistd.h>
RTPtimeshift.cc: char* being compared to 0 instead of the atoi() of it
rtp_payload_registry_unittest.cc: avoid narrowing int to uint32.
BUG=2700
R=andrew@webrtc.org , fischman@webrtc.org , henrik.lundin@webrtc.org , henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5019004
Patch from Victor Costan <costan@gmail.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5248 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-09 21:34:30 +00:00
c329529047
Apply transaction to setting connected to Room entities, to resolve a possible race condition at two clients connecting simultaneously.
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BUG = 1742
Test = Apprtc Integration Test
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5247 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-09 19:37:45 +00:00
70ddf9355f
libyuv r905 with yuv off by 1 fix for valgrind overread
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BUG=none
TEST=valgrind build bots
R=andrew@webrtc.org , mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5246 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-09 18:17:42 +00:00
de7c9e8884
Ensure WEBRTC_MODULE_UTILITY_VIDEO is undefined for enable_video==0.
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Move the logic to common.gypi to reduce the chance of the define being
unprotected in the future.
BUG=b/12018143
TESTED=git try, and local Linux build with -Denable_video=0
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5244 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-09 16:23:00 +00:00
