generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity
4,058 Commits 1 Branch 0 Tags 108 MiB
814e28413d
Commit Graph

4058 Commits

This Branch
This Branch
All Branches
Author SHA1 Message Date
mflodman@webrtc.org
1c986e7c89 Removed ViE file API. 
R=asapersson@webrtc.org, niklas.enbom@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1723004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4267 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-26 09:12:49 +00:00
solenberg@webrtc.org
a5fd2f1348 Do basic parsing of RTCP headers in PcapFileReader to enable log filtering. 
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1697004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4266 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-26 08:36:07 +00:00
solenberg@webrtc.org
892d750ba6 Add *.DS_Store to .gitignore so that ._.DS_Store is ignored too. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1698004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4265 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-26 08:22:53 +00:00
solenberg@webrtc.org
91811e2b04 Remove unused multi stream bandwidth estimator. 
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1712004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4264 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-25 20:36:14 +00:00
stefan@webrtc.org
a4c5abb52a Make sure padding packets are sent. 
BUG=1837
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1717006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4260 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-25 15:46:16 +00:00
vikasmarwaha@webrtc.org
bb25256775 Added changes in apprtc demo to ignore turn address through query string for FF. Also made sure that the iceServers array doesnot include transport parameter in turn url for FF. Finally removed turn: from the turn_url when creating iceservers for pre-M28 chrome. 
R=dutton@google.com, juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/1627006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4259 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-25 14:52:51 +00:00
sergeyu@chromium.org
3348ae2b97 mac: Mark kCGLPFAFullScreen as allowed in a 10.6-only function. 
kCGLPFAFullScreen is marked deprecated starting with 10.6 in the 10.9 SDK,
but it's functional on 10.6 and this code only runs on 10.6 and will go away
when support for 10.6 is dropped.

BUG=webrtc:1958
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1710004

Patch from Nico Weber <thakis@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4255 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-21 23:33:10 +00:00
marpan@webrtc.org
bb4f225a5b Roll libvpx to 207593. 
-pick up libvpx roll to c259af4f.

TBR: ajm@google.com

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1707004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4254 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-21 22:19:34 +00:00
hclam@chromium.org
6eb53f71d6 Fix memory bot failure 
Exit the method with critical setting held. This should make
the memory bot happy.

TBR=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1704005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4251 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-20 23:01:39 +00:00
hclam@chromium.org
2e402ce873 Enqueue packet in pacer if sending fails 
If a packet cannot be sent while pacer is in use it should be
queued. This avoid packet loss due to congestion.

BUG=1930
R=pwestin@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1693004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4250 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-20 20:18:31 +00:00
mikhal@webrtc.org
9ca7360b97 VCM: removing max jitter estimate 
BUG= 1921
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1690004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4249 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-20 20:13:07 +00:00
andrew@webrtc.org
0851df8d60 Remove unneeded *_NOT_SUPPORTED from VoEAudioProcessing. 
* Remove ANDROID_NOT_SUPPORTED from a bunch of echo metrics calls
where it actually is supported.
* No error to call GetTypingDetectionStatus.
* Consolidate typing detection disablement to reduce boilerplate.

R=niklas.enbom@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1683004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4247 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-19 17:03:47 +00:00
stefan@webrtc.org
8ccb9f9716 Fixes some pacer/padding issues found while testing. 
- A bug was introduced in r4234 causing no paced packets to be sent.
- Only update the sequence number counter if a padding packet is actually going to be sent, to avoid packet loss.
- Have all packets go through the pacer if pacing is enabled to avoid reordering.
- Fix race condition on reading capture_time_ms_/timestamp_ in rtp_sender.cc.

BUG=1837
TEST=trybots and vie_auto_test --automated
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1682004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4246 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-19 14:13:42 +00:00
kjellander@webrtc.org
2d7617afce Add dummy Android test APK to be used for buildbot automation testing. 
Until we have WebRTC test targets created for Android, this test
makes it possible to move forward for buildbot automation.

TEST=Android NDK buildbot and local execution of:
source build/android/envsetup.sh
gclient runhooks
ninjar -C out/Debug
verified the out/Debug/simple_apk dir exists and has the files.
BUG=1882
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1688005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4245 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-19 09:10:49 +00:00
fbarchard@google.com
d7148c86c5 Use 3 threads for higher than 720p resolutions 
BUG=1893
TEST=untested
R=ajm@google.com, andrew@webrtc.org, dingkai@google.com, marpan@google.com, marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1684004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4243 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-18 22:06:42 +00:00
hclam@chromium.org
30fb7b83d5 Add a log message to see video delay break down 
Shows video delay in terms of:
1. Min playout delay
2. Jitter delay
3. Max decode time
4. Render delay

R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1674004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4242 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-18 21:37:09 +00:00
kjellander@webrtc.org
6cfe178af2 Chromium Android tools for test execution. 
The md5sum and forwarder2 binaries from Chromium's
src/tools/android are needed to be able to run tests using the
test framework launched by build/android/run_tests.py.
Since they depend on Chromium's base, we're using a precompiled
copy for WebRTC's purposes.

Linux works out of the box if Chromium's Android build instructions
at https://code.google.com/p/chromium/wiki/AndroidBuildInstructions
are used. Mac runs into problems earlier in the build toolchain,
but as Mac is not a supported Android development platform in Chrome,
the files will have to be copied manually on that platform for now.

TEST=Synced, built and ran a test APK using run_tests.py.
BUG=1882
R=andrew@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1679005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4241 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-18 07:14:33 +00:00
sergeyu@chromium.org
a20eb91154 Make ScreenCapturerMac work in versions of OSX before Lion. 
The screen capturer was broken when moving code to webrtc: width
and height parameters for glReadPixels were swapped by mistkake.

BUG=crbug.com/244102
R=alexeypa@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1678005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4239 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-17 22:22:40 +00:00
sergeyu@chromium.org
9e182795a9 Enable ScreenCapturer unittests 
previously ScreenCapturer unittests were disabled by mistake

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4238 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-17 21:14:36 +00:00
sergeyu@chromium.org
a590b41c9a Use intptr_t to represent window IDs on all platforms. 
Previously void* was used on windows which makes it harder to work
with the IDs in cross-platform code.

R=alexeypa@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1672004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4237 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-17 20:02:21 +00:00
stefan@webrtc.org
508a84b255 Wire up pacer-based padding. 
This connects the pacer-based padding with the RTP modules, which will
generate padding packets roughly according to what the pacer suggests.
It will only generate padding packets of maximum size to keep the number
off padding packets as small as possible. This also sets a limit of how much
padding + media bitrate which the pacer is allowed to "request" from the
RTP modules.

Padding will for now only be generated by the first sending RTP module.

BUG=1837
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1612005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4234 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-17 12:53:37 +00:00
stefan@webrtc.org
50fb4afade Revert r4145 "Revert 4127 "Switch frame list implementation to std::map."" 
TBR=tnakamura@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1678004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4233 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-17 07:33:58 +00:00
stefan@webrtc.org
c8b29a2feb Revert r4146 "Revert 4104 "Refactor jitter buffer to use separate lists for de..."" 
TBR=tnakamura@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1677004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4232 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-17 07:13:16 +00:00
hclam@chromium.org
7262ad1385 Fix AV sync issue 
r4229 introduced an AV sync issue due to an error.
This is a one linear fix and provides the correct
current video delay for synchronization.

TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1675004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4231 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-15 06:51:27 +00:00
hclam@chromium.org
9b23ecb939 Log current and target AV delay in ViESyncModule 
R=mikhal@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1668006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4229 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-14 23:30:58 +00:00
kjellander@webrtc.org
63e988856e Merge more tests into modules_{unit,integration}tests. 
A new test target named 'modules_integrationtests' is created
and the following test targets were merged into it:
* audio_coding_module_test
* test_fec
* video_coding_integrationtests
* vp8_integrationtests

A couple of other targets were merged into modules_unittests:
* audio_coding_unittests
* audioproc_unittest
* common_unittests
* video_coding_unittests
* video_processing_unittests
* vp8_unittests

I wasn't able to merge audio_decoder_unittests and neteq_unittests due to
conflicts with different defines in these tests.

Some tests that have special requirements aren't merged into
modules_integrationtests yet. I took the opportunity to rename them
since the bot configs will need to be update anyway:
* audio_device_test_api -> audio_device_integrationtests
* video_capture_module_test -> video_capture_integrationtests
* video_render_module_test -> video_render_integrationtests

Exclude files were added for modules_integrationtests to make sure
the memcheck and tsan bots doesn't tests that are too slow
(audio_coding_module_test and vp8_integrationtests were previously
disabled on those bots).

Suppressions for AudioCodingModuleTest needed to be added to get
modules_integrationtests to pass memcheck (even if the test is
excluded from execution).

BUG=1843
TEST=local execution on Linux and trybots (passing except the merged tests of course)
R=andrew@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1656004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4228 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-14 20:09:44 +00:00
henrike@webrtc.org
f27389ca9f WebRTCDemo: ensures that using front and back camera work as expected. 
I.e. egress: Real world up is stream up.
Ingress: stream up is app up.
Local (preview): Real world up is app up.

BUG=1763
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1642004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4227 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-14 05:37:13 +00:00
henrike@webrtc.org
d4ed1a3e2c Fixes linker issue with no op trace. 
BUG=N/A
R=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4226 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-13 09:34:54 +00:00
braveyao@webrtc.org
a19333954d Apprtc CSS: Add flip to local view of FireFox and remove warning of Canary 
BUG=1380
TEST=Manual Test
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/1620004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4225 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-13 03:49:03 +00:00
turaj@webrtc.org
fee739c224 Risk of division by zero. 
bug=b9338699

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1634004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4223 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-12 20:10:06 +00:00
fischman@webrtc.org
dd97ef4e28 Revert 4211 "Build all java files into jar for each module on An..." 
Reason for revert: behold the meltdown of the "trunk" bots on http://build.chromium.org/p/chromium.webrtc.fyi/waterfall

Turns out that include in gyp is fraught with peril: https://code.google.com/p/gyp/wiki/InputFormatReference#Including_Other_Files

> Build all java files into jar for each module on Android
>
> BUG=
> R=fischman@webrtc.org, niklas.enbom@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1636004
>
> Patch from Jeremy Mao <yujie.mao@intel.com>.

TBR=fischman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/1660005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4222 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-12 17:39:29 +00:00
kjellander@webrtc.org
20a993f88a Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test. 
Take two of http://review.webrtc.org/1657004/
This time with execution on trybots.

BUG=1925
TEST=win,win_rel,mac,mac_rel,linux,linux_rel trybots passing.
R=mflodman
TBR=mflodman

Review URL: https://webrtc-codereview.appspot.com/1658004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4221 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-12 14:38:01 +00:00
kjellander@webrtc.org
935d705370 Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test. 
Disable on Windows due to failures on bots.

BUG=1925
TEST=compile on Linux and Windows.
R=mflodman
TBR=mflodman

Review URL: https://webrtc-codereview.appspot.com/1657004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4220 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-12 13:59:57 +00:00
kjellander@webrtc.org
04996cd5e5 Fix breakage due to test_fec conversion to gtest. 
In my attempt to commit a subset of http://review.webrtc.org/1647005/
instead of all of it, I forgot to add the gtest dependency to the
test_fec.gypi. This CL fixes that.

TEST=local compile + win_rel,mac_rel,linux_rel trybots
BUG=1916
R=marpan
TBR=marpan

Review URL: https://webrtc-codereview.appspot.com/1655004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4219 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-12 12:15:33 +00:00
kjellander@webrtc.org
22bbbdfa68 Convert test_fec to gtest 
All tests needs to be gtest tests in order to be executed
with the upcoming isolate/swarm framework.

TEST=trybots passing
BUG=1916
R=andrew@webrtc.org, marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/1647005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4218 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-12 11:55:05 +00:00
kjellander@webrtc.org
7124dd8561 Disable ViEStandardIntegrationTest.RunsRtpRtcpTestWithoutErrors test. 
BUG=1790
TEST=Just local compilation.
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1654004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4217 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-12 08:28:09 +00:00
kjellander@webrtc.org
18275a8429 Update bots to make LKGR progress. 
This is just a temporary fix until we have fixed a working solution for
the new buildbot waterfalls in Chrome infrastructure.

TEST=none
BUG=none
R=phoglund
TBR=phoglund

Review URL: https://webrtc-codereview.appspot.com/1654005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4216 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-12 08:10:18 +00:00
tina.legrand@webrtc.org
b097670264 G722_1/G722_1C codecs won't instantiate 
BUG=issue1890
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1650004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4215 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-12 07:41:42 +00:00
fbarchard@google.com
2ef9513916 libyuv r723 with convert util -attenuate feature used to fix transparent pixels used by Effects. By attenuating and then unattenuating, any transparent pixels will have RGB value of black, which will filter correctly when bilinear resized. 
BUG=none
TEST=try bots
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1652004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4214 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-11 22:03:29 +00:00
kjellander@webrtc.org
6c35e0b0f7 Reorganize test targets in WebRTC 
This CL will lower the number of test targets in WebRTC by:

Add common_audio_unittests and merge the following targets into it (copied from http://review.webrtc.org/1584006):
* resampler_unittests
* signal_processing_unittests
* vad_unittests

Merge into modules_unittests:
* bitrate_controller_unittests
* desktop_capture_unittests
* media_file_unittests
* remote_bitrate_estimator_unittests
* rtp_rtcp_unittests
* paced_sender_unittests

Merge into test_support_unittests:
* channel_transport_unittests

channel_transport.gyp was also removed in favor for test.gyp.

I had to remove a main method from rtcp_format_remb_unittest.cc
since it caused the fileutils.h code to not be able to find the
right project root path in ordrer to provide correct paths
to test files.

Buildbot configuration update will be synced with the commit of this CL.

TEST=trybots
BUG=1843
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4213 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-11 08:29:17 +00:00
kjellander@webrtc.org
6d6d95e2b8 Add support for test disable files in webrtc_tests.py 
Adding support for text files in
tools/valgrind-webrtc/gtest_exclude that are used by the
wrapper script for memory tool execution (webrtc_tests.py).

This allows fine-grained disabling of tests using checked in
text files instead of maintaining such in the buildbot config.

For more details on naming of these text files and what to put
in them, see:
http://www.chromium.org/developers/tree-sheriffs/sheriff-details-chromium/memory-sheriff#TOC-Excluding-tests

TEST=local execution of tsan and memcheck on Linux, using an
exclude file (done during development of http://review.webrtc.org/1647005)
BUG=none
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1648004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4212 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-11 06:03:32 +00:00
fischman@webrtc.org
1374965680 Build all java files into jar for each module on Android 
BUG=
R=fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1636004

Patch from Jeremy Mao <yujie.mao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4211 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-10 23:34:27 +00:00
alexeypa@chromium.org
4af0878e57 Allow the screen capturer to capture oversized cursors and cursors without alpha channel (Windows). 
Changes in this CL:
  - CaptureCursor() scans the cursor to verify that it has alpha channel.
  - The AND mask of the cursor is used to reconstruct transparency if the cursor does not have alpha channel.
  - CaptureCursor() always outlines the cursor when a "screen reverse" pixel detected.  Previously it was only done for black and while cursors.
    
Added desktop_capture_unittest.MouseCursorShapeTest to test the cursor conversion code.
    
BUG=chromium:223147
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1627004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4210 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-10 22:29:17 +00:00
alexeypa@chromium.org
5e03f8ab67 Landing binary cursor image files to be used in a follow up CL. 
See https://webrtc-codereview.appspot.com/1627004/ for more details. TBR since that CL has been reviewed and LGTMed.

TBR=sergeyu@chromium.org

BUG=chromium:223147

Review URL: https://webrtc-codereview.appspot.com/1647004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4209 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-10 21:07:31 +00:00
fbarchard@google.com
dfa1c4afc6 libyuv r722 for OWNERS file for chromium, white space fix for lint, unittests on scale use randomize to reduce overhead, and neon change from vld1.u8 to vld1.8 for better compiler portability. 
BUG=none
TEST=none
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1643005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4207 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-10 19:35:17 +00:00
fischman@webrtc.org
fe6b57187d AppRTCDemo: delete hosted android_channel.html now that it's no longer necessary. 
This file became redundant with 47220050 which rolled to libjingle opensource in
r327 of talk/examples/android/src/org/appspot/apprtc/GAEChannelClient.java

R=vikasmarwaha@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1606004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4206 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-10 17:22:50 +00:00
elham@webrtc.org
5137b9752f Updated WebRTC version to 3.33 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1645004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4204 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-10 17:03:51 +00:00
mflodman@webrtc.org
509754c4c9 Making no NACK mode work again in VideoEngine. 
BUG=1910
TEST=ViE autotest loopback with no protection and some percent packet loss
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1631004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4203 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-10 15:50:12 +00:00
pbos@webrtc.org
1819fd711a RW lock access to ssrc maps in VideoCall. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1640004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4202 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-10 13:48:26 +00:00
solenberg@webrtc.org
adb51f5709 Add back the WEBRTC_DIRECT_TRACE flag. 
BUG=
R=andresp@webrtc.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1596004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4201 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-10 09:03:41 +00:00