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Author SHA1 Message Date
pbos@webrtc.org
3ff788cf73 Increase timeout for AsyncWriteTest.TestWrite. 
Having a 10ms timeout for something meant to run on DrMemory is insane.

TBR=henrike@webrtc.org
BUG=3490

Review URL: https://webrtc-codereview.appspot.com/23959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7410 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-09 12:47:15 +00:00
kwiberg@webrtc.org
4bd2db9a55 Opus wrapper: Use const for inputs and uint8[] for byte streams 
About half of the functions already followed the desired pattern; this
patch fixes the other half.

BUG=909
R=aluebs@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7409 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-09 11:21:10 +00:00
kjellander@webrtc.org
1bada48401 Make DEPS find check_root_dir.py in legacy checkouts. 
In r7405 the DEPS hook wasn't properly handling the case
when the trunk dir is not yet renamed. This makes the script
only be called if it exists in the old not-yet-renamed trunk dir.

BUG=3534
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7408 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-09 10:53:02 +00:00
minyue@webrtc.org
2c0cdbce22 Estimating NTP time with a given RTT. 
RemoteNtpTimeEstimator needed user to give a remote SSRC and it intended to call RtpRtcp module to obtain RTT, to be able to calculate Ntp time.

When RTT cannot be directly obtained from the RtpRtcp module with the specified SSRC, RemoteNtpTimeEstimator would fail.

This change allows RemoteNtpTimeEstimator to calculate NTP with an external RTT estimate.

An immediate benefit is that capture_start_ntp_time_ms_ can be obtained in a Google hangout call.

BUG=

TEST=chromium + hangout call
R=stefan@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7407 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-09 10:52:43 +00:00
minyue@webrtc.org
c803907d87 Removing useless packets when inserting them (NetEq) 
This is to save the buffer.

Some old code may become unnecessary, and will be removed in a separate CL.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7406 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-09 10:49:54 +00:00
kjellander@webrtc.org
0b0ac8236b Remove root_dir variable from DEPS + enforce rename. 
Update DEPS to no longer have the root_dir variable.
Add a script that detects if the user have a solution named
'trunk' and explains what needs to be done to change it to 'src'.

The reason for this change is that bot_update doesn't support
custom_vars in solutions and Chrome infra is trying to get
rid of them entirely in the future.

The bots are already using a solution named 'src' so they
won't run into this error during runhooks.

BUG=3534
TESTED=Ran the script with a .gclient containing a solution
named 'trunk' and one named 'src'. Also tested the presence
of the custom_vars dict and not.

R=andrew@webrtc.org, hinoka@chromium.org

Review URL: https://webrtc-codereview.appspot.com/30619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7405 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-09 09:11:27 +00:00
bjornv@webrtc.org
3ea35fdb1b common_audio: Removed macro WEBRTC_SPL_LSHIFT_W16 
The macro was a trivial << operation and where used has been replaced by <<. Affected components are
* ilbc
* isacfix

BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22919005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7404 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-09 08:47:02 +00:00
pbos@webrtc.org
127ca3f8e5 Disable TestDTLSConnectWithSmallMtu on all platforms. 
Other bots elsewhere are breaking on this test, my money is on that this
might be due to different SSL versions being used on the different bots.
This test fails on at least a couple of bots that has use_openssl=1.

R=kjellander@webrtc.org
TBR=henrike@webrtc.org
BUG=3910

Review URL: https://webrtc-codereview.appspot.com/25839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7403 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-09 07:52:03 +00:00
andrew@webrtc.org
0001adcfef Use openmax_dl on all ARM (v7 or higher) platforms. 
openmax_dl now works on non-Android ARM, but it still requires
arm_version >= 7, and doesn't work on iOS at all.

TEST=Chromium build for a ChromeOS ARM device passes.
BUG=chromium:415393
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7402 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-09 04:13:02 +00:00
glaznev@webrtc.org
95bacfed08 Remove bad waiting code from video decoder release function. 
Instead keep surface texture object alive while video codec
is re-initialized with a different resolution.

BUG=
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7401 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-09 00:00:11 +00:00
buildbot@webrtc.org
97abeee282 (Auto)update libjingle 77263371-> 77296420 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7400 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-08 22:24:30 +00:00
henrike@webrtc.org
536eb98408 Re-enables a bunch of base unittests. 
BUG=3836
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7399 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-08 22:17:02 +00:00
andrew@webrtc.org
9ea539605e Roll chromium_revision fc668e2..2d714fa (298195:298667) 
Picks up openmax_dl fixes for non-Android ARM.

Summary of changes (git diff fc668e2..2d714fa DEPS):
* third_party/boringssl c7dd5f3..51fcd87
* third_party/openmax_dl/dl/src 79e64bc..0164270
* third_party/usrsctp/usrsctplib d5685d4..dfd687b
* tools/swarming_client 33d442a..c28b74f

TBR=kjellander
BUG=chromium:415393,webrtc:3906

Review URL: https://webrtc-codereview.appspot.com/23929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7398 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-08 19:16:10 +00:00
andrew@webrtc.org
4165f7aa22 Add a variable for deciding when to use openmax_dl. 
Modifies the previous condition to additionally not use openmax_dl on
iOS. Remove the All target's direct dependency on it, as it is now
pulled in by the targets that need it.

Add gn support since an openmax_dl gn target is available.

BUG=chromium:415393, webrtc:3906
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7397 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-08 18:01:27 +00:00
bjornv@webrtc.org
f71785cd3b audio_coding: Replaced macro WEBRTC_SPL_RSHIFT_W16 with >> 
Replaced trivial shift macro with >>. The actual implementation of the macro is simply >>.

Affected codecs:
* ilbc
* isac/fix

BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7396 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-08 15:36:30 +00:00
pbos@webrtc.org
575d126a3d Protect send_/recv_streams_ in WebRtcVideoEngine2. 
Important as OnLoadUpdate() won't be called on the worker thread and the
list of streams can't be concurrently modified while delivering this
callback to all send streams.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/22959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7395 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-08 14:48:08 +00:00
kwiberg@webrtc.org
9c6dc46c6d CHECK/DCHECK: Explicitly state whether the condition can have side effects 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7394 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-08 12:19:56 +00:00
henrik.lundin@webrtc.org
5e3d7c78de Change name of a NetEq internal member variable 
In the StatisticsCalculator class, the member last_report_timestamp_
was unfortunately named. It's now been changed to
timestamps_since_last_report_, which describes it more accurately.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7393 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-08 12:10:53 +00:00
jiayl@webrtc.org
742922b313 Make the media content send only if offerToReceive is false while local streams exist. 
We previously do not add the media content if offerToReceive is false.

BUG=3833
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7390 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-07 21:32:43 +00:00
pbos@webrtc.org
d6bda09503 Initialize sctp_paddrparams in OpenSctpSocket(). 
Addresses 'use-of-uninitialized-value' detected with MemorySanitizer.
params.spp_address.sa_family was used without being initialized before
when calling usrsctp_setsockopt with SCTP_PEER_ADDR_PARAMS.

R=jiayl@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/23909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7389 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-07 19:23:43 +00:00
pbos@webrtc.org
27e5898f45 Explicitly unpoison FDs for MSan. 
MSan doesn't handle inline assembly that's used by FD_ZERO causing a
false positive.

R=earthdok@chromium.org, henrike@webrtc.org
BUG=chromium:344505

Review URL: https://webrtc-codereview.appspot.com/25799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7388 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-07 17:56:53 +00:00
glaznev@webrtc.org
46ffc70878 Temporary fix to allow Invoke() calls for VP8 HW encoder and decoder. 
BUG=
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7387 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-07 17:11:36 +00:00
pbos@webrtc.org
963b979510 Remove potential deadlock in WebRtcVideoEngine2. 
Fixes lock-order inversions between capturer's SignalVideoFrame and
WebRtcVideoSendStream. Additionally also removes all deadlock
suppressions for WebRtcVideoEngine2.

R=stefan@webrtc.org
TBR=kjellander@webrtc.org
BUG=1788,2999

Review URL: https://webrtc-codereview.appspot.com/26729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7386 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-07 14:27:27 +00:00
kjellander@webrtc.org
a9e363e721 Roll chromium_revision c264a05..fc668e2 (297113:298195) 
Mainly to pick up https://codereview.chromium.org/619723006/
to fix our MSan bot.

Summary of changes (git diff c264a05..fc668e2 DEPS):
* third_party/boringssl 01fe820..c7dd5f30
* third_party/usrsctp/usrsctplib 8975bd5..d5685d4
* tools/swarming_client 79940aee..33d442a

Clang updated 216630:217949 (git diff c264a05..fc668e2 tools/clang/scripts/update.sh)
This caused TSan v2 to hit an assert in libjingle_peerconnection_unittest
and libjingle_media_unittest, which is why the dlclose call
had to be disabled for now (webrtc:3895).

BUG=3895
R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7385 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-07 12:49:34 +00:00
pbos@webrtc.org
77d5a57e5c Revert "Only configure the SSL library in one place." 
This reverts commit r7378, which broke Windows compile targets
elsewhere.

R=kjellander@webrtc.org
TBR=henrike@webrtc.org
BUG=chromium:413497

Review URL: https://webrtc-codereview.appspot.com/28679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7384 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-07 11:43:03 +00:00
kjellander@webrtc.org
6ed1cf49f0 Isolate: Remove use of --ignore_broken_items 
BUG=chromium:395700
R=jam@chromium.org

Review URL: https://webrtc-codereview.appspot.com/30659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7383 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-07 09:17:35 +00:00
henrik.lundin@webrtc.org
9103953b58 Fix neteq_rtpplay so that empty SSRC is valid 
In r7380, the command line flag --ssrc was added to neteq_rtpplay.
However, it was not possible to omit that flag, since the validation
did not accept an empty string. This CL fixes that.

TBR=kwiberg@webrtc.org
BUG=2692

Review URL: https://webrtc-codereview.appspot.com/24869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7382 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-07 07:18:36 +00:00
henrik.lundin@webrtc.org
7cbc4f969a Set NetEq playout mode through the Config struct 
This change opens up the possibility to set the playout mode when
creating the NetEq object. The old methods SetPlayoutMode and
PlayoutMode are still available, but are deprecated.

BUG=3520
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7381 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-07 06:37:39 +00:00
henrik.lundin@webrtc.org
8b65d511a0 Add an SSRC filter to neteq_rtpplay 
This allows the user to set the desired SSRC if the input file
contains multiple streams.

BUG=2692
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7380 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-07 05:30:04 +00:00
turaj@webrtc.org
532ed43e85 Prevent reading outside iSAC bitstream, if the stream is corrupted. 
BUG=chrome_373312(#24)
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7379 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-07 00:21:02 +00:00
henrike@webrtc.org
8234fa6f0e Only configure the SSL library in one place. 
Build settings now use use_openssl in both Chromium and standalone builds. It
moves all the platform-specific SSL-related build checks to be conditioned on
this flag as appropriate.

This is to avoid colliding with Chromium's transition away from NSS.

BUG=chromium:413497
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7378 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-06 22:30:46 +00:00
henrike@webrtc.org
2fe5893748 Mac: adds missing _DEBUG flag to mac debug builds. 
BUG=3836
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7377 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-06 22:04:11 +00:00
henrike@webrtc.org
528fc650d8 Fixing build issue with L-sdk 
Based on https://codereview.appspot.com/153000043/

BUG=https://code.google.com/p/chromium/issues/detail?id=420293
R=niklas.enbom@webrtc.org, serya@chromium.org, yfriedman@chromium.org

Review URL: https://webrtc-codereview.appspot.com/29659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7374 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-06 17:56:43 +00:00
henrike@webrtc.org
9a742b4840 talk: removes empty directories base and sound. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7373 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-06 17:52:59 +00:00
houssainy@google.com
5d3e7ac1a3 Check on the existence of report directory 
Reports will be written at rtcBot/test/reports/<report_name>

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7372 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-06 17:21:27 +00:00
pbos@webrtc.org
42684be21b Wire up CPU adaptation in WebRtcVideoEngine2. 
Includes clean-up work to be able to use the webrtc::Call::Config that's
set up. This introduced a CallFactory to spawn a FakeCall with the
config used and allowed removal of FakeWebRtcVideoChannel2.

BUG=1788
R=mflodman@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7370 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-03 11:25:45 +00:00
henrike@webrtc.org
31b75eae05 Moves xmllite's unittests to rtc_unittest. 
BUG=3836
R=andrew@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26669005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7369 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-02 18:43:47 +00:00
glaznev@webrtc.org
25cc745d6b Switch to SW video decoder on Android after getting 2 or more 
critical errors from HW decoder.

BUG=410730
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7368 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-02 16:58:05 +00:00
henrik.lundin@webrtc.org
4b133da5fd Let RtpFileSource use RtpFileReader 
RtpFileSource used to implement it's own RTP dump file reader, but
with this change it simply uses RtpFileReader. One benefit is that
pcap files are now also supported.

All NetEq test tools that use RtpFileSource are updated.

BUG=2692
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7367 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-02 08:19:38 +00:00
bjornv@webrtc.org
348eac641e audio_processing: Replaced WEBRTC_SPL_RSHIFT_U32 with >> 
A trivial macro that is replaced. Affected components:
* AGC
* NSx

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29599005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7366 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-02 08:07:05 +00:00
sergeyu@chromium.org
5fa8c458d8 Remove mouse cursor capturer from the ScreenCapturer interface 
Mouse can be captured using MouseCursorMonitor and all code in chromium
already uses it instead of ScreenCapturer.

R=jiayl@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7363

Review URL: https://webrtc-codereview.appspot.com/31529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7365 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-02 01:47:10 +00:00
sergeyu@chromium.org
6138f0f89d Revert "Remove mouse cursor capturer from the ScreenCapturer interface" 
This reverts commit 0adc4953512ee0a57cf7f3c0591b024c2316554a. It broke
FYI bots

TBR=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/27649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7364 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-02 01:36:43 +00:00
sergeyu@chromium.org
1fced0f2aa Remove mouse cursor capturer from the ScreenCapturer interface 
Mouse can be captured using MouseCursorMonitor and all code in chromium
already uses it instead of ScreenCapturer.

R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7363 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-02 00:18:10 +00:00
sergeyu@chromium.org
76819d315d Add error trap for XFixesGetCursorImage() 
BUG=https://code.google.com/p/webrtc/issues/detail?id=3245
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7362 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 23:07:12 +00:00
andrew@webrtc.org
325cff01b4 Import LappedTransform and friends. 
Add code for doing block-based frequency domain processing. Developed
and reviewed in isolation. Corresponding export CL:
https://chromereviews.googleplex.com/95187013/

R=bercic@google.com, kjellander@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7359 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 17:42:18 +00:00
henrike@webrtc.org
593c3a0868 rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation. 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7358 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 16:33:03 +00:00
henrike@webrtc.org
4530b2ca48 Revert 7355 "Fix parallelization in libjingle_p2p_unittest." 
Breaks waterfall.

TBR=pbos@webrtc.org
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/22909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7357 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 15:43:55 +00:00
henrike@webrtc.org
36b0c1afae Adds PRESUBMIT.py dispensation for depending on rtc_base. 
Dispensation for: a few test suites, desktop capture and libjingle.

BUG=N/A
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7356 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 14:40:58 +00:00
pbos@webrtc.org
fd29205e6e Fix parallelization in libjingle_p2p_unittest. 
Adding VirtualSocketServers to SessionTest and RelayServerTest to avoid
contention on real ports.

R=juberti@webrtc.org
BUG=2597
TEST=third_party/gtest-parallel/gtest-parallel -w 64 out/Debug/libjingle_p2p_unittest

Review URL: https://webrtc-codereview.appspot.com/26679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7355 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 12:31:31 +00:00
pbos@webrtc.org
c86e45d7c4 Fix parallelizability in modules_tests. 
R=henrik.lundin@webrtc.org
BUG=3873
TEST=third_party/gtest-parallel/gtest-parallel -r 10 -w 64 out/Debug/modules_tests

Review URL: https://webrtc-codereview.appspot.com/24799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7354 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 10:05:40 +00:00