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Author SHA1 Message Date
pbos@webrtc.org
eb7b7bce3d Modify video_render/ to allow a single old frame. 
This stabilizes tests as a single frame reaches end-to-end, as well as
allowing slow or heavily-loaded systems to see any video updates even if
the frame takes more than 500ms in the pipeline.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=2724

Review URL: https://webrtc-codereview.appspot.com/5949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5303 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-16 18:24:37 +00:00
fischman@webrtc.org
5b3c67ef25 objc/README: Remove outdated advice about target_os. 
BUG=chromium:248168
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5979005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5302 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-16 17:15:19 +00:00
pbos@webrtc.org
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
asapersson@webrtc.org
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
bjornv@webrtc.org
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
pbos@webrtc.org
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
bjornv@webrtc.org
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
sprang@webrtc.org
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
turaj@webrtc.org
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
solenberg@webrtc.org
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
turaj@webrtc.org
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
stefan@webrtc.org
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
turaj@webrtc.org
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
wu@webrtc.org
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
mflodman@webrtc.org
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
asapersson@webrtc.org
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
pbos@webrtc.org
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
pbos@webrtc.org
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
mallinath@webrtc.org
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
sprang@webrtc.org
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
mflodman@webrtc.org
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
henrik.lundin@webrtc.org
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
wu@webrtc.org
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
turaj@webrtc.org
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
wu@webrtc.org
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
wu@webrtc.org
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
henrike@webrtc.org
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
asapersson@webrtc.org
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
fischman@webrtc.org
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
henrike@webrtc.org
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
fischman@webrtc.org
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
wu@webrtc.org
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
pbos@webrtc.org
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
pbos@webrtc.org
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
turaj@webrtc.org
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
hta@webrtc.org
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
kjellander@webrtc.org
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
sprang@webrtc.org
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
kjellander@webrtc.org
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
sprang@webrtc.org
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
hta@webrtc.org
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
hta@webrtc.org
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
hta@webrtc.org
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
andrew@webrtc.org
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
fischman@webrtc.org
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
andrew@webrtc.org
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
sprang@webrtc.org
8b8819262f Improve VideoSendStreamTest::MaxPacketSize 
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
kjellander@webrtc.org
917306d3fd Change uses of the obsolete armv7 setting to arm_version==7. 
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
fischman@webrtc.org
eb7def234e Fix compilation errors on Fedora 20. 
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
braveyao@webrtc.org
c329529047 Apply transaction to setting connected to Room entities, to resolve a possible race condition at two clients connecting simultaneously. 
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