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Author SHA1 Message Date
pbos@webrtc.org
d60d79a145 Thread annotation of rtc::CriticalSection. 
Effectively re-lands r5516 which was reverted because talk/-only
checkouts existed. This now resides in webrtc/base/, so no talk/-only
checkouts should be possible.

This change also enables -Wthread-safety for talk/ and fixes a bug in
talk/media/webrtc/webrtcvideoengine2.cc where a guarded variable was
read without taking the corresponding lock.

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

Review URL: https://webrtc-codereview.appspot.com/27569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7284 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-24 07:10:57 +00:00
pbos@webrtc.org
38344ed280 Move thread_annotations.h to webrtc/base/. 
R=andresp@webrtc.org, mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/27579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7283 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-24 06:05:00 +00:00
glaznev@webrtc.org
8166faeff3 Change Android video renderer to maintain video aspect 
ratio when displaying camera or decoded video frames.

-

R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7282 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 23:58:52 +00:00
glaznev@webrtc.org
90668b1633 Switch HW video decoder to output byte buffers if video 
renderer EGL context is not provided by app.

R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7281 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 21:42:15 +00:00
buildbot@webrtc.org
1b7dcc1647 (Auto)update libjingle 76169599-> 76176062 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7280 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 17:41:48 +00:00
johannkoenig@google.com
94ff92ceec Use VPX_IMG_FMT_*/VPX_PLANE_* defines 
The compatibility layer has been removed upstream:
https://gerrit.chromium.org/gerrit/gitweb?p=webm%2Flibvpx.git;a=commit;h=9cdaa3d72eade9ad162ef8f78a93bd8f85c6de10

BUG=webrtc:3839
R=marpan@google.com, marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7279 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 17:31:47 +00:00
guoweis@webrtc.org
2c1bcea1bc Enable ipv6 by default for webrtc under a Finch experiment. 
Reapply 23529005 after fixing the build break issue (Chromium:582133002)

Committed: https://code.google.com/p/webrtc/source/detail?r=7253

Review URL: https://webrtc-codereview.appspot.com/23529005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7278 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 16:23:02 +00:00
henrik.lundin@webrtc.org
3987f10c11 Revert "Remove DTMF status methods from Voice Engine" r7276 
This change caused some trouble.

TBR=henrika@webrtc.org,pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7277 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 13:15:14 +00:00
henrik.lundin@webrtc.org
bf7b9e0081 Remove DTMF status methods from Voice Engine 
These methods are not used.

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

Review URL: https://webrtc-codereview.appspot.com/24689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7276 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 12:54:04 +00:00
kjellander@webrtc.org
e34a2e7475 Revert "Set minimum SDK level to 10.7 for Mac and iOS" (r7175) 
Reverting this since it didn't fix the build failures.
We ended up passing mac_sdk=10.9 in GYP_DEFINES on the bots
to to make the build pass again
(https://codereview.chromium.org/573673002).

BUG=3120
R=mcasas@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7275 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 12:43:14 +00:00
pbos@webrtc.org
faf2410a32 gn: Hide modules/video_capture:video_capture_internal_impl behind an arg 
R=andresp@webrtc.org, brettw@chromium.org, kjellander@webrtc.org, pbos@webrtc.org, brettw

Review URL: https://webrtc-codereview.appspot.com/30479004

Patch from Cem Kocagil <ckocagil@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7274 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 12:37:06 +00:00
henrik.lundin@webrtc.org
0e6e4d2ff2 Reland "Converting five tests to use new AudioCoding interface" (r7258) 
This CL reverts r7264. The problem was that iSAC-SWB and iSAC-FB are
not supported on android. These are now disabled.

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7273 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 12:05:34 +00:00
andresp@webrtc.org
4f6f22f0c6 Reland (rev 7259) "Convert AcmReceiverTest to new AudioCoding interface" 
Was reverted by mistake in 7260. Actual culprit was 7258.

BUG=3520
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7272 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 11:37:57 +00:00
bjornv@webrtc.org
ea29787df0 audio_processing/agc: Solved building with AGC_DEBUG + few style changes 
webrtc did not build if AGC_DEBUG was turned on. This CL fixes that. Has no impact on performance since it is development/debug code.

* Name change to WEBRT_AGC_DEBUG_DUMP
* Added build flag agc_debug_dump to .gypi
* Added missing "%d" in printf at two places
* Some line length related style changes

Tested audioproc and modules_unittests with GYP_DEFINES=agc_debug_dump=1 webrtc/build/gyp_webrtc

BUG=N/A
TESTED=locally and trybots
R=aluebs@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7271 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 11:21:39 +00:00
pbos@webrtc.org
0a2087a711 Skeleton for registering external encoders/decoders. 
R=pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/31429005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7270 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 09:40:22 +00:00
tkchin@webrtc.org
c569a49a3d Unit tests for SSLAdapter 
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17309004

Patch from Manish Jethani <manish.jethani@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7269 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 05:56:44 +00:00
bjornv@webrtc.org
dc0b37dcb1 modules_unittests: Turned on ApmTest.Process test for Android 
The reason why ApmTest.Process breaks on Android is that two metrics over counts. I decided to add an offset and a different slack to the EXPECT_NEAR() calls that are affected. I think this is a reasonable approach since we have no more than two failing metrics. If any feature change that will make another metric fail, we should go back to the desk and find another way of solving this.

BUG=114
TESTED=locally on Nexus 7 and trybots
R=aluebs@webrtc.org, andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7268 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 05:03:44 +00:00
andrew@webrtc.org
a3c4d4dd2c Revert 7266 "WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type en..." 
This was causing apparently legitimate failures on the following bots:
http://chromegw/i/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/2599
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28KK%20Nexus5%29%28dbg%29/builds/2023
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28JB%20Nexus7.2%29%28dbg%29/builds/1825
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28KK%20Nexus5%29/builds/2013
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28JB%20Nexus7.2%29/builds/1795

> WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t
> 
> We have to fix both at once, since there's a macro that calls one of
> them or the other.
> 
> BUG=909
> R=andrew@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, minyue@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/19229004

TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7267 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 01:32:57 +00:00
kwiberg@webrtc.org
8c5740b485 WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t 
We have to fix both at once, since there's a macro that calls one of
them or the other.

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

Review URL: https://webrtc-codereview.appspot.com/19229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7266 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-22 23:04:14 +00:00
pbos@webrtc.org
83f95ba9a6 Remove engine-level SetOptions. 
Already removed in WebRtcVideoEngine.

R=andresp@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/29549005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7265 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-22 16:07:18 +00:00
andresp@webrtc.org
99e404c84a Revert "Converting five tests to use new AudioCoding interface" (rev 7258). 
This time reverts the Cl that actually broke the tests. Got the wrong rev before. :/

BUG=3520
TESTED=Locally with CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py gtest -s modules_unittests --gtest_filter=AcmReceiverBitExactness.8kHzOutput --verbose --isolate-file-path=webrtc/modules/modules_unittests.isolate
TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7264 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-22 15:49:56 +00:00
houssainy@google.com
35850ff71f Adding test file path as argument of the rtcBot run command's arguments. 
The new command to run rtcBot is:-
node test.js <bot_type> <test_file_path>

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7263 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-22 15:24:56 +00:00
henrik.lundin@webrtc.org
64a2f10f4b Remove Get/SetNetEQPlayoutMode APIs 
These are not used anymore.

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

Review URL: https://webrtc-codereview.appspot.com/26569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7262 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-22 14:30:10 +00:00
houssainy@google.com
07ca949346 Adding webrtc_video_streaming test 
This test is streaming video and audio between two bots using webrtc js api.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7261 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-22 13:52:39 +00:00
andresp@webrtc.org
c570761288 Revert "Convert AcmReceiverTest to new AudioCoding interface" (rev 7258). 
Breaks android modules_unittests tests by crashing on AcmReceiverBitExactness.8kHzOutput
Was already visible on "git cl try" before submitting on https://webrtc-codereview.appspot.com/23719004/#

BUG=3520
R=kwiberg@webrtc.org, henrik.lundin@webrtc.org
TBR=kwiberg@webrtc.org, henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7260 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-22 13:18:34 +00:00
henrik.lundin@webrtc.org
cfe073539c Convert AcmReceiverTest to new AudioCoding interface 
In order to maintain test coverage for the old API (AudioCodingModule)
during the transition period, the old test was copied to
AcmReceiverTestOldApi.

Modified and extended AudioCoding and the implementation to make the
test compile and run.

Created a converter method from new to old config struct

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7259 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-22 12:10:44 +00:00
henrik.lundin@webrtc.org
eb1de5cb72 Converting five tests to use new AudioCoding interface 
The converted tests are:
AcmIsacMtTest
AcmReceiverBitExactness
AcmSenderBitExactness
AudioCodingModuleMtTest
AudioCodingModuleTest

In order to maintain test coverage for the old API (AudioCodingModule)
during the transition period, the old tests were copied and given the
suffix OldApi:
AcmIsacMtTestOldApi
AcmReceiverBitExactnessOldApi
AcmSenderBitExactnessOldApi
AudioCodingModuleMtTestOldApi
AudioCodingModuleTestOldApi

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7258 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-22 12:07:12 +00:00
aluebs@webrtc.org
bdfdc96b22 Clang-format ns_core 
BUG=webrtc:3811
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7257 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-22 10:59:46 +00:00
pbos@webrtc.org
759982d357 Set number of temporal layers for VideoSendStream. 
Introduces a mapping between EncoderConfig and VideoCodec. More
specifically it also removes an assert that there should be no set
temporal layers in the new API, which is wrong and was temporary.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/25619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7256 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-22 09:32:46 +00:00
henrik.lundin@webrtc.org
612171527e Ensure that NetEq recovers after a large timestamp jump 
Before this change it could happen that a large jump in timestamp (a
jump not correlated to wall-clock change) caused the audio to go silent
without recovering. The reason was that all incoming packets after the
jump were considered too old compared to the last decoded packet, and
were deleted. With CL changes two things:

1. If the only available packet in the buffer is an old packet, NetEq
will do Expand instead of immediate reset. This is to avoid that one
late packet triggers a reset.

2. Old packets are discarded only when the decision to decode a packet
has been taken. This is to allow the buffer to grow and eventually
flush if no decodable packet has been found for some time.

This CL also includes a new unit test for this situation.

BUG=3785
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7255 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-22 08:30:07 +00:00
henrike@webrtc.org
88772874da Disabled several rtc_unittests so the tests can be turned on in the waterfall 
BUG=3836
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7254 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-22 07:30:48 +00:00
guoweis@webrtc.org
97ed39344a Reapply 23529005 after fixing the build break issue (Chromium:582133002) 
Review URL: https://webrtc-codereview.appspot.com/23529005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7253 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 21:06:12 +00:00
buildbot@webrtc.org
ed5ca1f122 (Auto)update libjingle 75925673-> 75926712 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7252 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 20:30:44 +00:00
buildbot@webrtc.org
c98f217c65 (Auto)update libjingle 75924589-> 75925673 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7251 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 20:18:10 +00:00
buildbot@webrtc.org
0c9fe72b21 (Auto)update libjingle 75922684-> 75924589 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7250 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 20:05:02 +00:00
glaznev@webrtc.org
ebf2757339 Fix HW video decoder crash on some Android KK devices. 
Remove direct access to decoder Java output buffer memory
when HW decoder is configured to decode to surface.

-

R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30459005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7249 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 19:36:13 +00:00
thorcarpenter@google.com
c1eebfa107 Fix the libjingle_media_unittest failure in Windows build by modifying libjingle_tests.gyp and sctpdataengine_unittests.cc instead of ssladapter.cc. 
R=harryjin@google.com, pthatcher@webrtc.org, tpsiaki@google.com

Review URL: https://webrtc-codereview.appspot.com/22699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7245 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 17:54:00 +00:00
glaznev@webrtc.org
e65812427d Fixing compilation failure in peerconnection_jni.cc with WEBRTC_CHROMIUM_BUILD. 
Symbol LogcatTraceContext not defined.
Submitting on behalf of serya@.
Dup of https://webrtc-codereview.appspot.com/29529004/

TEST=Build target libjingle_peerconnection_javalib with applied CL https://codereview.chromium.org/551793003/
BUG=https://crbug.com/383418
R=serya@chromium.org

Review URL: https://webrtc-codereview.appspot.com/28529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7244 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 16:53:46 +00:00
aluebs@webrtc.org
fbf3bfe172 Separate between Analyze and Process in NS 
Filled the empty analyze API, separating the noise estimation from the process API.
No formatting fixes or extra refactoring has been done, to make the review process easier.
This patch has been tested for bit-exactness over the whole QA set in every aggressiveness.

BUG=webrtc:3811
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7243 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 15:18:59 +00:00
kjellander@webrtc.org
95705602bd Additional disabled tests in rtc_unittests. 
It appears https://review.webrtc.org/27559004/
not enough to get rtc_unittests up and running.
It's currently failing on Linux 32, Linux ASan
and Win SyzyASan bots.

BUG=3836
TBR=henrike@webrtc.org
TEST=Locally passing rtc_unittests on Linux Release
build with asan=1 and lsan=1 in GYP_DEFINES.

Review URL: https://webrtc-codereview.appspot.com/24659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7242 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 14:49:37 +00:00
kjellander@webrtc.org
34ac7762e0 Additional disabled tests in rtc_unittests. 
It appears https://review.webrtc.org/30449004 was
not enough to get rtc_unittests up and running.

BUG=3836
TEST=Locally passing rtc_unittests on Mac Debug.
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7241 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 13:47:47 +00:00
henrike@webrtc.org
fded02c164 base: disabled several base tests on Mac so that rtc_unittests can be turned back on 
BUG=N/A
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7240 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 13:10:10 +00:00
pbos@webrtc.org
bbe0a8517d Config struct for VideoEncoder. 
Used for config parameters in common between multiple codecs as well as
the encoder-specific pointer. In particular this contains content mode
(realtime video vs. screenshare).

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

Review URL: https://webrtc-codereview.appspot.com/16319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7239 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 12:30:25 +00:00
andresp@webrtc.org
02686115cc Re-enable missing android tests disabled due to issue 3770. 
BUG=3770
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7238 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 08:24:19 +00:00
andresp@webrtc.org
2036a7bb40 Clean directx_sdk_path as it is already defined in base/common.gypi 
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7237 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 08:14:12 +00:00
henrik.lundin@webrtc.org
5ca6008236 Creating a test helper class TimestampJumpRtpGenerator 
This class provides a way to test with an RTP sequence that make an
arbitrary jump in the timestamp series.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7236 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 07:14:31 +00:00
buildbot@webrtc.org
6e5c78422d (Auto)update libjingle 75875619-> 75878731 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7235 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 06:46:37 +00:00
buildbot@webrtc.org
b5a5c44ef7 (Auto)update libjingle 75865376-> 75875619 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7234 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 05:36:18 +00:00
buildbot@webrtc.org
d7acf11e8d (Auto)update libjingle 75854833-> 75865376 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7233 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 02:01:09 +00:00
buildbot@webrtc.org
ccb3e3f3db (Auto)update libjingle 75854418-> 75854833 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7232 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-18 23:31:03 +00:00