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Author SHA1 Message Date
pkasting@chromium.org
4591fbd09f Use size_t more consistently for packet/payload lengths. 
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.

This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.

BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom

Review URL: https://webrtc-codereview.appspot.com/23129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-20 22:28:14 +00:00
pbos@webrtc.org
ece3890d3a Report total bitrate for all streams in GetStats. 
This regression wasn't caught because I accidentally disabled multiple
streams for EndToEndTest.GetStats in a refactoring.

R=stefan@webrtc.org, xians@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/27179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7701 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-14 11:52:04 +00:00
marpan@webrtc.org
4765ca55f9 Roll chromium_revision: 28d1981..d3db2ff 
Pick up the libvpx roll: https://codereview.chromium.org/674753002

Summary of changes (28d1981..d3db2ff/DEPS):
* third_party/android_tools 36bf7ac..ea50ccc
* third_party/boringssl 7ea8481..751e889
* third_party/icu 8ac906f..d8b2a9d
* third_party/libvpx efe9712..2e5ced5
* third_party/usrsctp/usrsctplib
* tools/gyp 1990:1991
* tools/swarming_client a57d7db..bcb3bc3

Clang is not updated in this roll.

Made the change getchar() --> getc(stdin) as seems like getchar() isn't supported on android anymore.
(getchar() was causing the error: undefined reference to '__srget')

Update rate control parameter in vp9 test.

R=andrew@webrtc.org
TBR=ajm@google.com

Review URL: https://webrtc-codereview.appspot.com/23229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7598 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-03 20:10:26 +00:00
marpan@webrtc.org
5b88317820 Add VP9 codec to VCM and vie_auto_test. 
Include VP9 tests in videoprocessor_integrationtests.
Include end-to-end send/receiveVP9 test.

This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422, which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in:
see https://code.google.com/p/webrtc/issues/detail?id=3932

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

Review URL: https://webrtc-codereview.appspot.com/31829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-01 06:10:48 +00:00
henrike@webrtc.org
b1dac33cac Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..." 
BUG=3932
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/27779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7470 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 18:54:46 +00:00
marpan@webrtc.org
573c78e31c Add VP9 codec to VCM and vie_auto_test. 
Include VP9 tests in videoprocessor_integrationtests.
Include end-to-end send/receiveVP9 test.
Passes trybots.

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

Review URL: https://webrtc-codereview.appspot.com/29449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7422 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-10 16:44:47 +00:00
andresp@webrtc.org
85ef770d92 Split video engine android initialization into each internal module initialization. 
This is to later on allow targets to pick at link time if to include the external or internal implementation. In order to do that the video_engine cannot compile different based on which option is picked later on.

BUG=3768,3770
R=glaznev@webrtc.org, stefan@webrtc.org
TBR=henrike@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7208 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-17 11:44:51 +00:00
asapersson@webrtc.org
9d453931c5 Change return value for number of discarded packets to be int. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7054 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 07:07:44 +00:00
pbos@webrtc.org
047a46f8b4 Remove Android.mk build files. 
These files are generally not maintained and break, some contain files
that don't exist anymore and do not build anymore. If we need to add
some of these back we should really set up a bot for them.

R=andrew@webrtc.org, glaznev@webrtc.org, henrike@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/15249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6974 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-26 08:48:51 +00:00
henrike@webrtc.org
6ac22e6b47 Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798 
R=andrew@webrtc.org, fbarchard@chromium.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6867 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-11 21:06:30 +00:00
pbos@webrtc.org
bc73871251 Remove the VPM denoiser. 
The VPM denoiser give bad results, is slow and has not been used in
practice. Instead we use the VP8 denoiser. Testing this denoiser takes
up a lot of runtime on linux_memcheck (about 4 minutes) which we can do
without.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6688 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 09:50:40 +00:00
stefan@webrtc.org
b9f5453e29 Add boilerplate code for H.264. 
R=mflodman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17849005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6603 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-04 12:42:07 +00:00
stefan@webrtc.org
cb254aac3b Enable pacing by default and remove the option to disable it from the new API. 
BUG=1672
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6416 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-12 15:12:25 +00:00
fischman@webrtc.org
b273b60154 ViEAutoTestAndroid: Unbreak compile by casting void* to jobject. 
Sure would be nice if the try fleet used both gcc _and_ clang...

TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6355 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 18:59:30 +00:00
fischman@webrtc.org
9512719569 AppRTCDemo(android): support app (UI) & capture rotation. 
Now app UI rotates as the device orientation changes, and the captured stream
tries to maintain real-world-up, matching Chrome/Android and Hangouts/Android
behavior.

BUG=2432
R=glaznev@webrtc.org, henrike@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15689005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6354 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 18:40:44 +00:00
fischman@webrtc.org
440e1d1053 vie_autotest_android.cc: stop referring to undefined functions. 
The roll in r6240 exposed the fact that vie_autotest_android.cc has been
depending on vie_autotest_network.cc since forever, even though that file isn't
part of the build!  #if'ing the references out to green the build.

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17599005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6241 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 21:40:45 +00:00
andresp@webrtc.org
60015d27ae Wire up --force_fieldtrials for vie_auto_test and for test targets linking with test/test.gyp:{test_main|test_support_main} 
This allows use of webrtc field trials  and opens up the possibility to try the different code paths when running the unit tests by wiring them up to a --force_fieldtrials.

Tested: running a test target that links with the above with a flag --force_fieldtrials=invalid leads the test to crash.

BUG=crbug/367114
R=mflodman@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6181 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-16 09:39:51 +00:00
kjellander@webrtc.org
98c76a120d Make vie/voe_auto_test accept non-supported flags without error. 
With the switch recipes on the buildbots and the deprecation of
the custom script at
https://code.google.com/p/webrtc/source/browse/trunk/webrtc/test/buildbot_tests.py
these tests will start failing when Chromium's runtest.py is passing
--brave-new-test-launcher --test-launcher-bot-mode
to the test.
A similar change was made for most of WebRTC's tests (that depends on
the test_support_main target) in
https://webrtc-codereview.appspot.com/2222005

BUG=chromium:346198
TEST=Successfully launched the executables on Linux and Mac using:
out/Release/voe_auto_test --brave-new-test-launcher --test-launcher-bot-mode --automated --test-launcher-summary-output=/tmp/tmpwhx6Zz
out/Release/vie_auto_test --brave-new-test-launcher --test-launcher-bot-mode --automated --capture_test_ensure_resolution_alignment_in_capture_device=false --test-launcher-summary-output=/tmp/tmpwhx6Zz

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

Review URL: https://webrtc-codereview.appspot.com/13499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6135 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 06:01:40 +00:00
mflodman@webrtc.org
f223746521 Upping start bitrate to min, if set to a lower value i SetSendCodec. 
BUG=3276
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21379005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6014 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 12:38:42 +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
mflodman@webrtc.org
5574dacd1f Log Fixit for parts of video_engine folder. 
BUG=3153
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5853 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-07 10:56:31 +00:00
solenberg@webrtc.org
4e65602886 Add API to allow deducting bitrate from incoming estimates before the capacity is distributed among outgoing video streams. For example, this can be used to reserve space for audio streams. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5791 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-26 14:32:47 +00:00
solenberg@webrtc.org
fc320466d1 Remove ViE external encryption API. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8079005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5525 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-11 15:27:49 +00:00
sprang@webrtc.org
7f73280ded Fraction lost statistics not being reported 
A bug is causing fraction lost to always be set to zero when calling
ViERTP_RTCP::Get(Send|Receive)ChannelRtcpStatistics. Fix this and update
tests to catch it.

BUG=
R=holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5235 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-06 11:56:55 +00:00
stefan@webrtc.org
0a3c1471b8 Add API to query video engine for the send-side delay. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4559005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5225 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-05 14:05:07 +00:00
henrik.lundin@webrtc.org
9fe3603dc1 Renaming ViEEncoderObserver::VideoSuspended 
New name is ViEEncoderObserver::SuspendChange.

BUG=2436
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5157 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 23:00:40 +00:00
braveyao@webrtc.org
425e1d0fb9 Typo in vie_autotest_win.cc 
BUG=2637
TEST=AutoTest
R=mflodman@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5145 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 02:17:01 +00:00
sprang@webrtc.org
dc50aaeaa8 Interface changes to old api, for use by new api transition. 
BUG=2589
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5142 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-20 16:47:07 +00:00
henrik.lundin@webrtc.org
ce8e0936d9 Rename AutoMute to SuspendBelowMinBitrate 
Changes all instances throughout the WebRTC stack.

BUG=2436
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5130 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-18 12:18:43 +00:00
kjellander@webrtc.org
e8722856f9 Disable all vie_auto_tests on Linux for now (take 2) 
Turns out OS_LINUX is not working in this context
(see http://review.webrtc.org/3539005/)
WEBRTC_LINUX is the right define to use.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5119 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-13 15:51:49 +00:00
kjellander@webrtc.org
c8489852ec Disable all automated vie_auto_tests on Linux for now 
Since the switch from icewm to openbox window manager on
Linux in Chrome infra, causes the test to hang when
creating Windows.

TEST=trybots compile step
BUG=chromium:318760
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3539005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5118 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-13 15:44:54 +00:00
fischman@webrtc.org
37bb4974e7 Expose VideoCodingModule's decoder stats up the stack from VCMTiming to chrome://webrtc-internals. 
R=juberti@google.com, mikhal@webrtc.org, stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5027 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-23 23:59:45 +00:00
henrik.lundin@webrtc.org
0d19ed9a06 AutoMute: Adding channel_id parameter to callback. 
BUG=2436
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2390004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5006 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-21 12:37:13 +00:00
fischman@webrtc.org
4e65e07e41 VideoCaptureAndroid: rewrote the (standalone) implementation of video capture on Android. 
Besides being ~40% the size of the previous implementation, this makes it so
that VideoCaptureAndroid can stop and restart capture, which is necessary to
support onPause/onResume reasonably on Android.

BUG=1407
R=henrike@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2334004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4915 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-03 18:23:13 +00:00
henrik.lundin@webrtc.org
7ea4f24ea5 Piping AutoMuter interface through to ViE API 
This is a piece of the AutoMuter effort. A second CL will follow containing modifications to the new API, and tests.

BUG=2436
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2331004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4899 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-02 13:34:26 +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
stefan@webrtc.org
cdd3d4d139 Revert test change in r4808. 
This was supposed to be an EXPECT_GT, I just misunderstood it in the previous CL. Added a sleep after the EXPECT_GT and before bytes_received_after = bytes_received_before.

BUG=1790
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2265006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4809 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-23 09:43:07 +00:00
stefan@webrtc.org
269dd4264f Reduce flakiness in network down test. 
The encoder is in the process of encoding when the network goes down, so we need to wait until it has finished before we expect no more packets to be sent.

Also fixed a test which was testing the wrong thing.

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2258008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4808 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-23 08:42:39 +00:00
stefan@webrtc.org
7a30dfdc69 Disable NACK bandwidth statistics test due to being too flaky. 
Tests for new API currently provide partial coverage, and will soon
provide full coverage.

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2151005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4789 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-19 12:08:55 +00:00
stefan@webrtc.org
b5a191bfe7 Fixes a flake in network down tests. 
And reduces the flakiness in NACK tests.

TESTS=trybots
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2258004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4788 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-19 11:14:35 +00:00
stefan@webrtc.org
7bb8f02274 Adds support for combining RTX and FEC/RED. 
This is accomplished by breaking out RTX and FEC/RED functionality from the RTP module and keeping track of the base payload type, that is the payload type received when not receiving RTX.

Enables retransmissions over RTX by default in the loopback test.

BUG=1811
TESTS=voe/vie_auto_test --automated and trybots.
R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2154004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4692 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-06 13:40:11 +00:00
elham@webrtc.org
814e28413d Revert r4562 
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2117004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4623 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-26 23:21:03 +00:00
agalusza@google.com
b655985abd Added choice of decode error mode to loopback test. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1997004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4562 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-16 23:07:14 +00:00
wu@webrtc.org
822fbd8b68 Update talk to 50918584. 
Together with Stefan's http://review.webrtc.org/1960004/.

R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2048004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-15 23:38:54 +00:00
marpan@webrtc.org
62ecc20afb Revert r4539 "Disable racy part of RunsRtpRtcpTestWithoutErrors". 
Bot failures for Win32-Release and Linux64-Release.

TBR=pbos@webrtc.org.

Review URL: https://webrtc-codereview.appspot.com/2026004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4541 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-13 21:36:48 +00:00
pbos@webrtc.org
a05653b2c1 Disable racy part of RunsRtpRtcpTestWithoutErrors. 
Disabled part as suggested in bug 1790, but without breaking it up into
multiple tests. These tests will be made redundant by tests for the new
API, and it would take far too long to clean these up properly.

BUG=1790
R=kjellander@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2022004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4539 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-13 14:27:20 +00:00
wu@webrtc.org
9dba525627 * Update libjingle to 50389769. 
* Together with "Add texture support for i420 video frame." from
wuchengli@chromium.org.
https://webrtc-codereview.appspot.com/1413004

RISK=P1
TESTED=try bots
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1967004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4489 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-05 20:36:57 +00:00
pbos@webrtc.org
12dc1a38ca Switch C++-style C headers with their C equivalents. 
The C++ headers define the C functions within the std:: namespace, but
we mainly don't use the std:: namespace for C functions. Therefore we
should include the C headers.

BUG=1833
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1917004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4486 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-05 16:22:53 +00:00
tnakamura@webrtc.org
aa4d96a134 Revert r4301 
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4357 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-16 19:25:04 +00:00