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Author SHA1 Message Date
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. 
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 
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. 
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. 
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
kjellander@webrtc.org
ffd242432d DrMemory: Suppress and exclude more tests to green up the full build. 
Exclude failing tests in libjingle tests and suppress
failures in modules_unittests.

Also exclude the following extremely slow tests to make it
possible to get a reasonable execution time for the bot:
* TestScaler.PointScaleTest (81573 ms)
* TestScaler.BiLinearScaleTest (1111554 ms)
* TestScaler.BoxScaleTest (1129625 ms)
* TestVp8Impl.BaseUnitTest (762598 ms)
* VideoProcessorIntegrationTest.ProcessNoLossChangeBitRate (342149 ms)

TBR=phoglund@webrtc.org
BUG=3158, 3183, 3184
TEST=Successful runs on local Windows box:
tools\valgrind-webrtc\webrtc_tests.bat --build-dir out --target Debug --test libjingle_p2p_unittest --tool drmemory_full
tools\valgrind-webrtc\webrtc_tests.bat --build-dir out --target Debug --test libjingle_peerconnection_unittest --tool drmemory_full
tools\valgrind-webrtc\webrtc_tests.bat --build-dir out --target Debug --test modules_unittests --tool drmemory_full

Review URL: https://webrtc-codereview.appspot.com/11819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5888 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-11 14:16:27 +00:00
stefan@webrtc.org
34c5da6b5e Cleaned up logging in video_coding. 
Converted all calls to WEBRTC_TRACE to LOG(). Also removed a large number of less useful logs.

BUG=3153
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5887 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-11 14:08:35 +00:00
asapersson@webrtc.org
8b2ec15d1e Convert WEBRTC_TRACE to LOG in utility. 
BUG=3153
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5886 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-11 07:59:43 +00:00
wu@webrtc.org
b884eb6118 (Auto)update libjingle 64630087-> 64709629 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5884 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-10 16:59:16 +00:00
henrike@webrtc.org
8dce41b3c6 Remove erronuous commit message from auto sync. 
BUG=N/A
TBR=kjellander@webrtc.org

http://webrtc-codereview.appspot.com/11639004/



git-svn-id: http://webrtc.googlecode.com/svn/trunk@5883 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-10 14:39:38 +00:00
pbos@webrtc.org
22cf7472a0 Disable UsesTraceCallback 
Ongoing removal of trace code is causing UsesTraceCallback to fail,
disabling it for now.

BUG=3157
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5882 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-10 14:39:22 +00:00
andresp@webrtc.org
e6013bb7be Fix loopback test for case where no constraint is given. 
R=stefan@webrtc.org
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5881 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-10 14:21:45 +00:00
asapersson@webrtc.org
2a770828d8 Remove usage of webrtc trace in video processing modules. 
BUG=3153
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11089005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5880 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-10 11:30:49 +00:00
andresp@webrtc.org
0273fa98e0 Add ability to control peer connection constraints for the loopback test. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11419005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5879 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-10 09:40:16 +00:00
buildbot@webrtc.org
15192f909e (Auto)update libjingle 64594651-> 64630087 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5878 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-10 06:34:32 +00:00