sergeyu@chromium.org
7d055a6e63
Revert r5937 "Fix multi-monitor support in the screen capturer for Mac."
...
This would break when rolled in chromium because some code in
chromium depends on the code I changed in that change.
TBR=jiayl@webrtc.org
BUG=361919
Review URL: https://webrtc-codereview.appspot.com/12199005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5940 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-18 23:45:38 +00:00
andrew@webrtc.org
0daa8be9d6
Add Chromium's ScopedVector.
...
Trivial changes from the original excepting scoped_vector_unittest.cc,
diff here: https://paste.googleplex.com/6664017300946944
This is a prerequisite for:
http://review.webrtc.org/9919004/
TBR=henrike@webrtc.org
BUG=2894
Review URL: https://webrtc-codereview.appspot.com/12179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5938 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-18 21:20:54 +00:00
sergeyu@chromium.org
be7585b150
Fix multi-monitor support in the screen capturer for Mac.
...
This feature was broken in r5471.
BUG=361919
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5937 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-18 18:22:41 +00:00
turaj@webrtc.org
a596a389ea
Fix iSAC/48000 issue with ACM2.
...
Registeration of iSAC into NetEq is through injecting and external AudioDecoder. This is because iSAC encoder and decoder need to share instances for bandwidth estimator to work. When external decoder is registerred, the sampling rate of the decoder had to be specified. iSAC/48000 decoder has a native sampling rate of 32000 Hz, but it has been registered as 48000 Hz decoder.
This CL fixing this issue by letting NetEq to obtain sampling rate of an external coder according to its existing database.
BUG=3143
TEST=voe_cmd_test,modules_unittest,try-bots
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5936 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 23:30:49 +00:00
fischman@webrtc.org
29540b1879
Revert "PeerConnectionFactory: delay deletion of owned threads."
...
This reverts r5933 because it broke
http://build.chromium.org/p/client.webrtc/builders/Win64%20Release/builds/1598
BUG=3100
Review URL: https://webrtc-codereview.appspot.com/12159004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5935 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 22:54:30 +00:00
buildbot@webrtc.org
1a87f529a2
(Auto)update libjingle 65151416-> 65151642
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5934 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 22:41:30 +00:00
fischman@webrtc.org
cea024d672
PeerConnectionFactory: delay deletion of owned threads.
...
Since PeerConnection holds a ref to its creating PeerConnectionFactory, it's
possible for ~PeerConnectionFactory() to be run on its signaling thread.
Deleting a thread from within that thread is sad times, so don't do it.
It would be nicer to avoid having PeerConnection hold a ref to the factory,
and instead require the user to keep the factory alive. Unfortunately that
changes the contract on PeerConnection{,Factory} and it's unclear how to vet
existing callers for safety.
BUG=3100
R=juberti@webrtc.org , noahric@google.com
Review URL: https://webrtc-codereview.appspot.com/11289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5933 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 22:36:00 +00:00
marpan@webrtc.org
b476d36120
Roll libvpx 259973:264320
...
TBR=ajm@google.com
Review URL: https://webrtc-codereview.appspot.com/12069007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5932 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 20:35:03 +00:00
henrike@webrtc.org
aeb0c28193
Update PRESUBMIT.py's list of "DO_NOT_SUBMIT_FILES".
...
BUG=N/A
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12029006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5931 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 14:15:43 +00:00
kwiberg@webrtc.org
e57ae02327
WebRtcAecm_Process: Reduce code duplication
...
BUG=
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5930 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 12:28:33 +00:00
kwiberg@webrtc.org
d2f366f28c
StereoToMono: Remove useless call to WebRtcSpl_SatW32ToW16
...
The max value is ((2**15 - 1) + (2**15 - 1)) >> 1
== (2**16 - 2) >> 1
== 2**15 - 1
which doesn't overflow.
The min value is (-2**15 + -2**15) >> 1
== -2**16 >> 1
== -2**15
which doesn't overflow.
Since those two bracket all possible results, the call to
WebRtcSpl_SatW32ToW16 is redundant.
BUG=
R=andrew@webrtc.org , bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12019004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5929 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 12:17:39 +00:00
henrika@webrtc.org
66803489f9
Removes parts of the VoEBase sub API as part of a clean-up operation where the goal is to remove unused APIs.
...
BUG=3206
R=henrik.lundin@webrtc.org , juberti@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12019005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5928 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 10:45:01 +00:00
henrika@webrtc.org
0f7375504a
Removes VoECodec sub API as part of a clean-up operation where the goal is to remove unused APIs.
...
BUG=3206
R=juberti@webrtc.org , niklas.enbom@webrtc.org , tina.legrand@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5927 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 10:38:08 +00:00
henrik.lundin@webrtc.org
e2e9abb3bc
Revert "Make VoiceEngine choose ACM2 by default"
...
The reason for reverting is that Issue 3143 should be resolved
first.
TBR=henrika@webrtc.org
BUG=3143
Review URL: https://webrtc-codereview.appspot.com/12119005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5926 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 10:12:27 +00:00
buildbot@webrtc.org
0b3c6c3191
(Auto)update libjingle 65086785-> 65104022
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5925 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 10:03:57 +00:00
henrik.lundin@webrtc.org
adaf809612
Removing AudioCoding duplicate tests
...
Reverting to using one version of ACM in ACM tests.
BUG=2996
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5924 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 08:29:10 +00:00
henrik.lundin@webrtc.org
6cec07f6a7
Make VoiceEngine choose ACM2 by default
...
The use of a factory for ACM will be removed in later CLs.
BUG=2996
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12069004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5923 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 08:07:18 +00:00
fischman@webrtc.org
c0a15b7ddc
Fix crashes due to dangling external decoder pointer.
...
When checking whether we need to release external decoder,
we have to do pointer comparison. We can't rely on payload
types, because payload types can be stale (e.g. before we
decode the first video frame after RegisterReceiveCodec).
This leaves a dangling pointer to external decoder, which
leads to crashes later, after we actually delete the
external decoder object.
This change has been verified in Chromecast code tree.
BUG=chromium:335539
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12049004
Patch from Sergey Volk <servolk@chromium.org>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5922 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 01:22:48 +00:00
buildbot@webrtc.org
39b868bad3
(Auto)update libjingle 65055925-> 65086785
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5921 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-17 00:04:39 +00:00
jiayl@webrtc.org
8f88f20af2
Expand the test max wait time from 1000ms to 2000ms.
...
The createOffer/createAnswer methods sometimes times out due to slow identity generation under memcheck.
BUG=2838
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5920 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-16 17:14:21 +00:00
kjellander@webrtc.org
c18729165a
Set include_internal_video_capture=1 for video_capture_tests
...
Having this override in the .gypi file avoids having to set it for the bots, which I think is best if we can avoid.
This CL also reverts r5869 so video_capture_tests are compiled for Android again.
BUG=2974,3152
TEST=Successfully ran:
git try -t compile
git try --bot=win_baremetal,mac_baremetal,linux_baremetal -t video_capture_tests
git try --bot=android_apk,android_apk_rel
Verified the change in webrtc/build/apk_tests.gyp makes the build compile successfully from a Chromium+WebRTC configured checkout for Android APK tests.
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5919 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-16 12:59:49 +00:00
aluebs@webrtc.org
f927fd6481
Re-enable AGC tests:
...
* AgcConfigTest.HasCorrectDefaultConfiguration
* AgcConfigTest.DealsWithInvalidParameters
* AgcConfigTest.CanGetAndSetAgcStatus
* AgcConfigTest.HasCorrectDefaultRxConfiguration
* AgcConfigTest.DealsWithInvalidRxParameters
* AgcConfigTest.CanGetAndSetRxAgcStatus
* AudioProcessingTest.AgcIsOnByDefault
* AudioProcessingTest.CanEnableAgcWithAllModes
* AudioProcessingTest.RxAgcShouldBeOffByDefault
* AudioProcessingTest.CanTurnOnDigitalRxAcg
* AudioProcessingTest.CannotTurnOnAdaptiveAnalogRxAgc
BUG=webrtc:2784
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12019006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5918 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-16 11:58:18 +00:00
kjellander@webrtc.org
7de47bce12
Remove use of tmpnam.
...
This solves compilation with the Mac SDK 10.9.
BUG=3120, 3151
TEST=git try -t modules_tests:VideoProcessorIntegrationTest*
R=fischman@webrtc.org , henrike@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10739005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5917 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-16 08:04:26 +00:00
andrew@webrtc.org
2c3f1abb69
Replace flooding logs in rtp_sender.cc with a comment.
...
Started occurring after:
https://webrtc-codereview.appspot.com/11129004
BUG=3153
R=andresp@webrtc.org , mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5916 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-15 21:26:34 +00:00
wu@webrtc.org
36eda7cf0e
Workaround for https://bugzilla.mozilla.org/show_bug.cgi?id=996329 , where the m line from firefox have a space at the end.
...
For example:
"m=application 38233 DTLS/SCTP 5000 "
BUG=3212
TEST=manually try to use DataChannel between FF 28 and Chrome with rtccopy.com
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12029005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5915 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-15 20:37:30 +00:00
fischman@webrtc.org
ca539bbed0
iOS: baby steps to being able to include_tests=1
...
- pull iossim in DEPS even when on mac (because bug 2152)
- fix audio_device_test_api.cc's use of bool instead of bool* (!)
- move unused-on-mobile message to non-mobile-only section of
hardware_before_streaming_test.cc
BUG=3185
R=kjellander@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5914 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-15 20:26:41 +00:00
henrik.lundin@webrtc.org
7c6e3d188a
Moved voe_neteq_stats_unittest to audio_coding_module_unittest
...
The design of VoeNetEqStatsTest in voice_engine_unittests depended on
being able to inject a factory for the audio coding module into
voice engine. This functionality is now likely going away, which would
make this test fail to compile. Further, the functionality under test
is mostly ACM functionality, wherefore it makes better sense to test it
at ACM level.
BUG=2996
R=henrika@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5912 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-15 17:59:25 +00:00
wu@webrtc.org
6c75c98964
Propagate capture ntp timestamp from rtp to renderer.
...
Mostly the interface changes, the real implementation of ntp timestamp will come in a follow up cl.
TEST=new tests and try bots
BUG=3111
R=niklas.enbom@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5911 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-15 17:46:33 +00:00
buildbot@webrtc.org
1fd5b45a0e
(Auto)update libjingle 64956819-> 64982143
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5910 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-15 17:39:43 +00:00
stefan@webrtc.org
2f8d5f3302
Check if a header extension is registered before updating it and fail silently if it's not.
...
BUG=
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5909 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-15 12:28:46 +00:00
kjellander@webrtc.org
190b72a350
Make libjingle Android example build without sourcing envsetup.sh
...
See https://webrtc-codereview.appspot.com/11799004
for full details (separate to avoid webrtc+talk changes in same CL).
BUG=chromium:346198
TEST=Local builds using:
. build/android/envsetup.sh
unset ANDROID_SDK_ROOT
webrtc/build/gyp_webrtc
ninja -C out/Debug
ninja -C out/Release
+ trybots passing: git try --bot=android,android_rel,android_clang
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5908 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-15 08:35:49 +00:00
kjellander@webrtc.org
6e105ede38
Make WebRTC Android examples build without sourcing envsetup.sh
...
The new recipes framework for configuring build explicitly sets the
GYP_DEFINES for Android builds instead of relying on the envsetup.sh script
which probably will be removed at some point in the future.
This causes our build to break since our Android examples relies on the
Android SDK being found using the ANDROID_SDK_ROOT environment variable.
A GYP variable 'android_sdk_root' exists and is set correctly by
common.gypi, which is what I'm using to pass this path correctly to these
tests.
The libjingle example is handled separately in
https://webrtc-codereview.appspot.com/11809004/
BUG=chromium:346198
TEST=Local builds using:
. build/android/envsetup.sh
unset ANDROID_SDK_ROOT
webrtc/build/gyp_webrtc
ninja -C out/Debug
ninja -C out/Release
+ trybots passing: git try --bot=android,android_rel,android_clang
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5907 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-15 08:35:00 +00:00
mallinath@webrtc.org
ad4440a64e
In shared socket mode, use udp port as default receiver even if
...
stun server address is not set.
This can happen in a loopback scenarios where clients do not need
to provide any server information.
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5906 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-15 01:10:58 +00:00
buildbot@webrtc.org
505f400f27
(Auto)update libjingle 64909599-> 64919255
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5905 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 20:33:47 +00:00
fischman@webrtc.org
e98598d3f0
Make everyone an OWNER for .gyp/.gypi add/delete purposes, talk/ edition.
...
This CL brought to you by:
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f;
done|sort|uniq|grep -v '^\.$'); do echo -e "\n# These are for the common case of
adding or renaming files. If you're doing\n# structural changes, please get a
review from a reviewer in this file.\nper-file *.gyp=*\nper-file *.gypi=*" >>
$d/OWNERS; done
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f;
done|sort|uniq|grep -v '^\.$'); do git add $d/OWNERS; done
(and then removed the non-talk/ impact)
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5904 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 20:31:16 +00:00
fischman@webrtc.org
2c89b5cb27
Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
...
This CL brought to you by:
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do echo -e "\n# These are for the common case of adding or renaming files. If you're doing\n# structural changes, please get a review from a reviewer in this file.\nper-file *.gyp=*\nper-file *.gypi=*" >> $d/OWNERS; done
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do git add $d/OWNERS; done
(and then removed the talk/ impact)
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5903 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 20:08:03 +00:00
henrik.lundin@webrtc.org
35ead381f8
Adding a config struct to NetEq
...
With this change, the parameters sent to the NetEq::Create method are
collected in one NetEq::Config struct. The benefit is that it is easier
to set, change and override default values, and easier to expand with
more parameters in the future.
BUG=3083
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5902 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 18:49:17 +00:00
henrik.lundin@webrtc.org
810acbc93e
New Packet and PacketSource classes for NetEq tests
...
These new classes are intended to replace the old NETEQTEST_RTPpacket
classes. The code in rtp_analyze.cc has been updated to use the new
classes; other test applications will follow.
BUG=2692
R=andrew@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5901 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 18:42:23 +00:00
buildbot@webrtc.org
1da6047132
(Auto)update libjingle 64813990-> 64909599
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5900 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 18:15:15 +00:00
fischman@webrtc.org
cf0b46c762
iosdeviceinfo.cc: remove unnecessary file
...
The do-nothing implementation in this file is already present in
mobiledevicemanager.cc (shared with Android) so this isn't adding value, and
causes duplicate-symbol errors under some compilers.
BUG=3201
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5899 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 18:12:32 +00:00
primiano@chromium.org
5cf73962e6
Fix gyp for video_capture/ensure_initialized.cc.
...
This is a follow-up to
https://webrtc-codereview.appspot.com/11359004
which introduced an invalid dependency in the
chromium build when building without linker GC.
BUG=2974,3152,chromium:354405
R=andrew@webrtc.org , fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11789005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5898 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 17:26:31 +00:00
buildbot@webrtc.org
f875f15afb
(Auto)update libjingle 64709629-> 64813990
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5897 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 16:06:21 +00:00
henrika@webrtc.org
b9309beea4
Removes VoECallReport sub API as part of a clean-up operation where the goal is to remove unused APIs.
...
BUG=3206
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5896 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 14:12:50 +00:00
xians@webrtc.org
5692531f18
Added a new OnMoreData() interface which will not feed the playout data to APM.
...
BUG=3147
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11059005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5895 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 10:50:37 +00:00
kjellander@webrtc.org
a956ec2019
Add win_drmemory_light trybot to default trybot list.
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BUG=chromium:360054
TEST=None
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5894 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 08:38:27 +00:00
kjellander@webrtc.org
940894d207
DrMemory: Excluding failing tests for Dr Memory Full
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PortAllocatorTest.TestEnableSharedSocketWithNat
fails in libjingle_p2p_unittest.
Example:
http://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Full/builds/2/steps/memory%20test%3A%20libjingle_p2p_unittest/logs/stdio
Previous experience tells me that disabling only this test
case will make another one in the same test suite fail,
so I'm blanked disabling the whole test.
TBR=phoglund
BUG=3158
Review URL: https://webrtc-codereview.appspot.com/11909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5893 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-13 11:45:43 +00:00
kjellander@webrtc.org
ef79fd7965
DrMemory: Excluding failing tests for Dr Memory Full
...
BUG=3158
TEST=None
TBR=kjellander,phoglund
Review URL: https://webrtc-codereview.appspot.com/11899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5892 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-13 08:31:42 +00:00
kjellander@webrtc.org
41e8774113
DrMemory: Excluding failing tests for Dr Memory Full
...
The PortTest.TestLocalToTurn of libjingle_p2p_unittest
and DtmfSenderTest.InsertDtmf of libjingle_peerconnection_unittest
failed in the first run on
http://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Full/
However, I cannot reproduce on my machine, so I'm disabling all
test cases of those tests, assuming the others might fail as well.
BUG=3158
TEST=None
TBR=phoglund
Review URL: https://webrtc-codereview.appspot.com/11889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5891 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-12 19:53:06 +00:00
jiayl@webrtc.org
8ce7c72456
Fix the captured screen rect conversion.
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device_mode.dmPosition is already relative to the primary display's top-left, while the expected value of GetScreenRect() is also relative to the primary display's top-left.
TESTED=verified on Windows single monitor capturing and cursor capturing is fixed.
BUG=https://code.google.com/p/chromium/issues/detail?id=362631
R=wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/11789006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5890 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-11 22:31:15 +00:00
turaj@webrtc.org
8d1cdaa84e
NetEq changes.
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BUG=
R=henrik.lundin@webrtc.org , minyue@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9859005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5889 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-11 18:47:55 +00:00