Commit Graph

4585 Commits

Author SHA1 Message Date
mallinath@webrtc.org
ad81ab8861 Suppress SSL error strings on mac_asan to unbreak that build
Example borkedness:
http://chromegw/i/client.webrtc/builders/Mac%20Asan/builds/642/steps/libjingl...

Original CL for this issue is here
https://webrtc-codereview.appspot.com/2263004/

and this got reverted in here
https://code.google.com/p/webrtc/source/diff?spec=svn4874&r=4872&format=side&path=/trunk/talk/base/openssladapter.cc&old_path=/trunk/talk/base/openssladapter.cc&old=4798

Trying to land it again now.

TBR=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2318005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4875 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-28 07:24:39 +00:00
wuchengli@chromium.org
30377c7f71 Change the parameters of calculating maximum decode time.
- Reduce the window size from 20 to 10 seconds. If there is
  any spurious high decode time, it will be faster to pass it.
- Ignore more samples at first because HW decoder has higher
  initialization latency.

BUG=crbug.com/298176
TEST=Run apprtc loopback on Chromebook Daisy.
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2315005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4874 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-28 06:06:18 +00:00
mallinath@webrtc.org
a27be8e4a1 Update libjingle to CL 53398036.
Review URL: https://webrtc-codereview.appspot.com/2323004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4872 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-27 23:04:10 +00:00
henrike@webrtc.org
34c50c1de1 Makes OpensSL default audio implementation/device on Android.
BUG=N/A
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2198005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4871 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-27 22:34:35 +00:00
kjellander@webrtc.org
a39b323a7e Add tools/sharding_supervisor to .gitignore
I should have thought about this in
http://review.webrtc.org/2314004/

BUG=none
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4870 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-27 20:20:47 +00:00
kjellander@webrtc.org
8b7ec82cf4 Exclude P2PTransportChannelSameNatTest.TestConesBehindSameCone for TSan Linux
I cannot reproduce locally but I can see 10+ consecutive
failing runs at the TSan RV bot at
http://build.chromium.org/p/client.webrtc.fyi/waterfall

TEST=verified the new exclude file is picked up by running:
tools/valgrind-webrtc/webrtc_tests.sh --tool=tsan_rv -b out/Release -t libjingle_p2p_unittest --gtest_filter=P2PTransportChannel*Test.*
(which runs the test if this file does not exist)
BUG=2396
TBR=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2320004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4869 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-27 20:03:03 +00:00
andrew@webrtc.org
4475905613 Disable flaky RapidSpeakerChange test.
Example:
chromegw/i/internal.client.webrtc/builders/Win32%20Debug/builds/762/steps/libjingle_p2p_unittest/logs/stdio

e:\b\build\slave\win32_debug\build\src\talk\session\media\currentspeakermonitor_unittest.cc(144):
error: Value of: kSsrc2
  Actual: 1002
Expected: current_speaker_
Which is: 1001
e:\b\build\slave\win32_debug\build\src\talk\session\media\currentspeakermonitor_unittest.cc(145):
error: Value of: 1
Expected: num_changes_
Which is: 2
[  FAILED  ] CurrentSpeakerMonitorTest.RapidSpeakerChange (16 ms)

TBR=wu@webrtc.org
BUG=2409

Review URL: https://webrtc-codereview.appspot.com/2318004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4867 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-27 18:24:40 +00:00
wu@webrtc.org
6049787252 Add protection to few more methods of AudioDeviceLinuxALSA. Those methods can be called from
a different thread.

One example is the _playWarning can be changed in AudioDeviceLinuxALSA::Init, which is called on the application's thread. At the same time it can be read via PlayoutWarning() on the VoE's process_thread.

RISK=P2
TESTED=try bots and tsan test:
tools/valgrind-webrtc/webrtc_tests.sh --tool=tsan -t out/Debug/libjingle_peerconnection_unittest --gtest_filter=PeerConnectionFactoryTestInternal.CreatePCUsingInternalModules
BUG=1205
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2315004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4866 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-27 18:19:25 +00:00
andrew@webrtc.org
137b3793d9 Only use -lm on Linux in ISAC.
Remove unneeded WEBRTC_LINUX define.

BUG=crbug.com/298656
TESTED=Passed trybots.
R=wjia@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2313004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4865 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-27 18:16:28 +00:00
kjellander@webrtc.org
287f07b174 Add sharding_supervisor to DEPS to prepare for swarm/isolated testing.
This component is now needed in addition to the swarm_client.

BUG=1916
TEST=I've run bots with this locally synced.
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2314004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4863 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-27 16:57:10 +00:00
pbos@webrtc.org
2e246b4e78 Remove test parameters from CallTest.
Since the test parameters weren't used, it made no sense to have a
parameterized test.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2316004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4862 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-27 10:54:10 +00:00
fischman@webrtc.org
3223a3d4ee Roll libvpx 212975:225010 to pick up iOS Release fixes
BUG=2332
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2305004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4861 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-26 15:59:48 +00:00
minyue@webrtc.org
663da0a8fc With ACM2 and NetEq4, VoE fuzz test very often fails.
As far as I observe, there are several reasons for this:
1. webrtc/modules/audio_coding/neteq4/neteq_impl.cc : 870
     assert(new_codec_);
   This is related to webrtc/modules/audio_coding/neteq4/decision_logic_normal.cc : 81
     kUndefined can happen without new_codec_ being set
2. webrtc/modules/audio_coding/neteq4/neteq_impl.cc : 745
   assert(sync_buffer_->FutureLength() >= expand_->overlap_length());
3. some other assert triggered.

The above happens not very often and goes away with no assertion.

3. (most common, this CL addresses this)
   webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc : 201
     payload_data_length = payload_length - rtp_header->header.paddingLength;
   There are situations that
     payload_length < rtp_header->header.paddingLength;
   OLD ACM + NetEq3 can handle this:
     a) webrtc/modules/audio_coding/main/source/acm_neteq.cc : 477
          int16_t payload_length = static_cast<int16_t>(length_payload);
        payload_length becomes negative in this situation
     b) webrtc/modules/audio_coding/neteq/recin.c
        WebRtcNetEQ_RecInInternal() handles negative payload length

I do not want to touch VoE, so I tried to let ACM2 and NetEq4 handle negative payload length.

This CL does not follow the exact way of OLD ACM + NetEq3. I stopped negative payload length at ACM and did not allow it go to NetEq4.

To try this, apply my uploaded patch : https://webrtc-codereview.appspot.com/2281004/

Let me know if you see better solutions.

R=henrik.lundin@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2292005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4860 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-26 15:21:26 +00:00
pbos@webrtc.org
f26e8f6f57 Remove include_dirs from tools.
BUG=1662
TEST=compile on trybots
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2306005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4859 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-26 14:04:27 +00:00
pbos@webrtc.org
f8b296628f Remove include_dirs from test.
BUG=1662
TEST=compile on trybots
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2304006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4858 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-26 14:03:57 +00:00
henrik.lundin@webrtc.org
544b17c6a9 Implemented AutoMuter in MediaOptimization
Also added a unittest. This is the first step towards creating an
AutoMuter function in WebRTC.

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

Review URL: https://webrtc-codereview.appspot.com/2294005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4857 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-26 12:05:15 +00:00
pbos@webrtc.org
04b61790d1 Remove include_dirs from pacing.
BUG=1662
TEST=compile on trybots
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2292006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4856 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-26 09:23:31 +00:00
pbos@webrtc.org
97eefb77e0 Remove include_dirs from remote_bitrate_estimator.
BUG=1662
TEST=compile on trybots
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2300004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4855 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-26 09:22:54 +00:00
pbos@webrtc.org
339fe12f67 Remove include_dirs from bitrate_controller.
BUG=1662
TEST=compile on trybots
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2301004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4854 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-26 09:22:34 +00:00
pbos@webrtc.org
054ccd2e35 Remove include_dirs from video_coding.
BUG=1662
TEST=compile on trybots
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2294007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4853 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-26 09:22:09 +00:00
pbos@webrtc.org
73f207611c Remove include_dirs from video_processing.
BUG=1662
TEST=compile on trybots
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2295005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4852 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-26 09:17:56 +00:00
pbos@webrtc.org
dc3fa08331 Remove include_dirs from rtp_rtcp.
BUG=1662
TEST=compile on trybots
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4851 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-26 09:17:13 +00:00
turaj@webrtc.org
7b75ac6756 Sync-packet insertion into NetEq4. This is related to r3883 & r4052 for NetEq 3.
BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4850 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-26 00:27:56 +00:00
andrew@webrtc.org
6b1e21924a Move the Config DelayCorrection struct to audio_processing.h.
TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/2304005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4849 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-25 23:46:20 +00:00
andrew@webrtc.org
1760a17b8e Add an extended filter mode to AEC.
Re-land: http://review.webrtc.org/2151007/
TBR=bjornv@webrtc.org

Original change description:
This mode extends the filter length from the current 48 ms to 128 ms.
It is runtime selectable which allows it to be enabled through
experiment. We reuse the DelayCorrection infrastructure to avoid having
to replumb everything up to libjingle.

Increases AEC complexity by ~50% on modern x86 CPUs.
Measurements (in percent of usage on one core):

Machine/CPU                                     Normal Extended
MacBook Retina (Early 2013),
Core i7 Ivy Bridge (2.7 GHz, hyperthreaded)     0.6%   0.9%

MacBook Air (Late 2010), Core 2 Duo (2.13 GHz)  1.4%   2.7%

Chromebook Pixel, Core i5 Ivy Bridge (1.8 GHz)  0.6%   1.0%

Samsung ARM Chromebook,
Samsung Exynos 5 Dual (1.7 GHz)                 3.2%   5.6%

The relative value is large of course but the absolute should be
acceptable in order to have a working AEC on some platforms.

Detailed changes to the algorithm:
- The filter length is changed from 48 to 128 ms. This comes with tuning
of several parameters: i) filter adaptation stepsize and error
threshold; ii) non-linear processing smoothing and overdrive.
- Option to ignore the reported delays on platforms which we deem
sufficiently unreliable. Currently this will be enabled in Chromium for
Mac.
- Faster startup times by removing the excessive "startup phase"
processing of reported delays.
- Much more conservative adjustments to the far-end read pointer. We
smooth the delay difference more heavily, and back off from the
difference more. Adjustments force a readaptation of the filter, so they
should be avoided except when really necessary.

Corresponds to these changes:
https://chromereviews.googleplex.com/9412014
https://chromereviews.googleplex.com/9514013
https://chromereviews.googleplex.com/9960013

BUG=454,827,1261

Review URL: https://webrtc-codereview.appspot.com/2295006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4848 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-25 23:17:38 +00:00
sergeyu@chromium.org
becbefaee6 Fix WindowCapturerWin to capture window decorations after window size changes.
DWM doesn't update window decorations in bitmap captured
from GetWindowDC(). Work it around by calling PrintWindow()
after each resize (which somehow affects what's later
captured from GetWindowDC()). That solution avoids the
downsides of PrintWindow() (namely flickering) while still
allowing to capture window decorations correctly.

BUG=crbug.com/289759
R=alexeypa@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2295004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4847 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-25 22:56:59 +00:00
turaj@webrtc.org
3fdeddb59a Disable a NetEq unittest on Android. The test tries to register iSAC-swb as send codec and fails.
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2303004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4845 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-25 22:19:22 +00:00
niklas.enbom@webrtc.org
3e7703640f Remove unused constants, so chrome can enable a warning for that. Patch from thakis@
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2296006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4844 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-25 22:05:05 +00:00
elham@webrtc.org
cecaae2e4c Updated WebRTC version to 3.43
TBR=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2296007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4842 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-25 21:45:23 +00:00
turaj@webrtc.org
0c0fae8a5e Re-enable verbose logging in NetEq4.
Using neteq4_speed_test there no complexity penalty is observed when verbose
logging is enabled.

BUG=2317
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2293004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4841 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-25 17:42:17 +00:00
fischman@webrtc.org
69fc315fd9 Convert DeviceInfoImpl::_captureCapabilities from a map to a vector.
The map was just mapping an index to a pointer of a POD, so the code is easily
simplified by using a vector (with implicit index key) and the POD as a value.
(also fixes a leak in the windows code, which lacked a virtual dtor for
VideoCaptureCapabilityWindows but was deleting through a base pointer).

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2298004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4840 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-25 17:01:42 +00:00
asapersson@webrtc.org
ce014d97cd Revert 4837 "Add an extended filter mode to AEC."
> Add an extended filter mode to AEC.
> 
> This mode extends the filter length from the current 48 ms to 128 ms.
> It is runtime selectable which allows it to be enabled through
> experiment. We reuse the DelayCorrection infrastructure to avoid having
> to replumb everything up to libjingle.
> 
> Increases AEC complexity by ~50% on modern x86 CPUs.
> Measurements (in percent of usage on one core):
> 
> Machine/CPU                                     Normal Extended
> MacBook Retina (Early 2013),
> Core i7 Ivy Bridge (2.7 GHz, hyperthreaded)     0.6%   0.9%
> 
> MacBook Air (Late 2010), Core 2 Duo (2.13 GHz)  1.4%   2.7%
> 
> Chromebook Pixel, Core i5 Ivy Bridge (1.8 GHz)  0.6%   1.0%
> 
> Samsung ARM Chromebook,
> Samsung Exynos 5 Dual (1.7 GHz)                 3.2%   5.6%
> 
> The relative value is large of course but the absolute should be
> acceptable in order to have a working AEC on some platforms.
> 
> Detailed changes to the algorithm:
> - The filter length is changed from 48 to 128 ms. This comes with tuning
> of several parameters: i) filter adaptation stepsize and error
> threshold; ii) non-linear processing smoothing and overdrive.
> - Option to ignore the reported delays on platforms which we deem
> sufficiently unreliable. Currently this will be enabled in Chromium for
> Mac.
> - Faster startup times by removing the excessive "startup phase"
> processing of reported delays.
> - Much more conservative adjustments to the far-end read pointer. We
> smooth the delay difference more heavily, and back off from the
> difference more. Adjustments force a readaptation of the filter, so they
> should be avoided except when really necessary.
> 
> Corresponds to these changes:
> https://chromereviews.googleplex.com/9412014
> https://chromereviews.googleplex.com/9514013
> https://chromereviews.googleplex.com/9960013
> 
> BUG=454,827,1261
> R=bjornv@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/2151007

TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2296005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4839 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-25 12:27:27 +00:00
andrew@webrtc.org
2f240b43f5 Disable some flaky libjingle base tests.
ThreadTest.Main and VirtualSocketServerTest.delay_v6

Example:
http://build.chromium.org/p/tryserver.webrtc/builders/win/builds/1234

TBR=wu@webrtc.org
BUG=2409

Review URL: https://webrtc-codereview.appspot.com/2297004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4838 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-25 02:33:50 +00:00
andrew@webrtc.org
26e02f0ee4 Add an extended filter mode to AEC.
This mode extends the filter length from the current 48 ms to 128 ms.
It is runtime selectable which allows it to be enabled through
experiment. We reuse the DelayCorrection infrastructure to avoid having
to replumb everything up to libjingle.

Increases AEC complexity by ~50% on modern x86 CPUs.
Measurements (in percent of usage on one core):

Machine/CPU                                     Normal Extended
MacBook Retina (Early 2013),
Core i7 Ivy Bridge (2.7 GHz, hyperthreaded)     0.6%   0.9%

MacBook Air (Late 2010), Core 2 Duo (2.13 GHz)  1.4%   2.7%

Chromebook Pixel, Core i5 Ivy Bridge (1.8 GHz)  0.6%   1.0%

Samsung ARM Chromebook,
Samsung Exynos 5 Dual (1.7 GHz)                 3.2%   5.6%

The relative value is large of course but the absolute should be
acceptable in order to have a working AEC on some platforms.

Detailed changes to the algorithm:
- The filter length is changed from 48 to 128 ms. This comes with tuning
of several parameters: i) filter adaptation stepsize and error
threshold; ii) non-linear processing smoothing and overdrive.
- Option to ignore the reported delays on platforms which we deem
sufficiently unreliable. Currently this will be enabled in Chromium for
Mac.
- Faster startup times by removing the excessive "startup phase"
processing of reported delays.
- Much more conservative adjustments to the far-end read pointer. We
smooth the delay difference more heavily, and back off from the
difference more. Adjustments force a readaptation of the filter, so they
should be avoided except when really necessary.

Corresponds to these changes:
https://chromereviews.googleplex.com/9412014
https://chromereviews.googleplex.com/9514013
https://chromereviews.googleplex.com/9960013

BUG=454,827,1261
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2151007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4837 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-25 02:17:47 +00:00
turaj@webrtc.org
d6a7a5f385 Small fixes to run ACM2 tests.
BUG=
R=minyue@google.com

Review URL: https://webrtc-codereview.appspot.com/2238004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4836 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-25 01:09:23 +00:00
turaj@webrtc.org
ff43c85ef1 API add to set background noise mode.
Background noise mode.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2194005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4835 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-25 00:07:27 +00:00
sergeyu@chromium.org
8d757ac0a2 Fix window capturer not to leak HDC.
Previously Windows window capturer kept DC handles between captures. As
result it was leaking DC handles in SelectWindow(). Fixed it so that it
calls GetWindowDC() for each capture.

R=alexeypa@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4834 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-24 23:13:51 +00:00
sergeyu@chromium.org
958cdf68f3 Fix window capturer to stop capturing when the target is minimized.
BUG=crbug.com/288205
R=alexeypa@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2273005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4833 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-24 22:10:13 +00:00
andrew@webrtc.org
f832a551cc Disable flaky TestPartialFrameHeader.
Example failure:
[http://chromegw/i/internal.client.webrtc/builders/Linux%20Asan/builds/657]

TBR=wu@webrtc.org
BUG=2409

Review URL: https://webrtc-codereview.appspot.com/2286004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4832 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-24 20:09:30 +00:00
andrew@webrtc.org
c2ac5ed588 Add more TSAN suppressions for libjingle_media_unittest.
BUG=2078
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2282004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4830 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-24 18:45:19 +00:00
andrew@webrtc.org
641587f938 Disable some VP8 tests on Android.
DecodeWithACompleteKeyFrame and FixedTemporalLayersStrategy.

TBR=andresp
BUG=2037

Review URL: https://webrtc-codereview.appspot.com/2283004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4829 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-24 18:43:28 +00:00
andrew@webrtc.org
f0f92fae12 Disable flaky SendDataMultipleClocks.
Example failure:
[http://chromegw/i/internal.client.webrtc/builders/Linux32%20Debug/builds/719]

TBR=mallinath
BUG=2409

Review URL: https://webrtc-codereview.appspot.com/2270005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4828 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-24 16:26:41 +00:00
henrika@webrtc.org
9b6eefcedf Fix for Heap-use-after-free in webrtc::voe::Channel::SendRTCPPacket
Ensures that we always call DeRegisterExternalTransport() even if a fuzz test calls DeRegisterExternalTransport in an unintialized state.

TBR=tommi
BUG=296804 in crbug.com

Review URL: https://webrtc-codereview.appspot.com/2275008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4827 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-24 15:42:49 +00:00
kjellander@webrtc.org
ebd3ab0ce1 Add libjingle_peerconnection_objc_test to buildbot_tests.py
To run this test, the executable in
libjingle_peerconnection_objc_test.app/Contents/MacOS/libjingle_peerconnection_objc_test
must be run. Running executables like this is not supported by
the buildbot scripts, which makes the paths to the stdio invalid.
Adding it to our command-line wrapping script and executing it as
'libjingle_peerconnection_objc_test' on the bots should work around
this limitation.

BUG=2120
TEST=locally building and executing the test on Mac using:
GYP_DEFINES="host_arch=x64 target_arch=x64" gclient runhooks
ninja -C out/Release
out/Release/buildbot_tests.py -t libjingle_peerconnection_objc_test

R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2275007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4826 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-24 13:41:16 +00:00
kjellander@webrtc.org
6e86349273 Disable tests that crash the OS X kernel when run under memcheck.
These libjingle tests crashes the OS X kernel when run under
memcheck on Mac 10.6. I didn't file bugs for them since it's unlikely
we can fix this anyway. There are several other tests disabled
in the libjingle code with similar comments, without bugs
assigned to them.
See talk/base/physicalsocketserver_unittest.cc for examples.

Affected waterfall: http://build.chromium.org/p/client.webrtc.fyi/waterfall

TEST=none
BUG=none
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2278006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4825 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-24 12:34:51 +00:00
stefan@webrtc.org
b0e6eb50b5 Revert r4823 "Reenable test and remove flaky expects."
TBR=mflodman@webrtc.org

BUG=2415

Review URL: https://webrtc-codereview.appspot.com/2277005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4824 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-24 10:38:57 +00:00
stefan@webrtc.org
01aad09a01 Reenable test and remove flaky expects.
BUG=2415
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2278005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4823 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-24 10:16:52 +00:00
henrik.lundin@webrtc.org
b426c469b9 MediaOptimization: Converting a few members to scoped_ptrs
For consistency with other parts of the code.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2275006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4822 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-24 07:41:53 +00:00
andrew@webrtc.org
6ffc74ee0e Disable flaky RunsRtpRtcpTestWithoutErrors.
TBR=mflodman
BUG=2415

Review URL: https://webrtc-codereview.appspot.com/2270006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4821 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-23 23:25:39 +00:00
andrew@webrtc.org
eb524d997b Remove deprecated AudioCodingModule::Destroy.
Have Channel hold a pointer rather than reference, and shorten the name.

TESTED=trybots
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2256004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4820 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-23 23:02:24 +00:00