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Author SHA1 Message Date
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
braveyao@webrtc.org
83a062cc5f AudioDeviceAndroidOpenSLES: NULL variables might be referenced in StopPlayout() 
BUG=1891
Test=ManualTest

R=fischman@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1622004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4200 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-10 08:09:05 +00:00
andrew@webrtc.org
569fdef732 Revert some variables to uint32_t to fix compile errors on Mac gcc. 
TBR=xians

Review URL: https://webrtc-codereview.appspot.com/1633004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4199 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-08 00:43:25 +00:00
andrew@webrtc.org
6f69eb78dd Allow audio devices with up to 64 channels on Mac. 
Does not increase memory requirements. Adds an additional check to ensure
configurations requiring more memory per IO block than the input ring buffer
contains are rejected.

BUG=1904
TESTED=Using Soundflower (64 channels) at 48 kHz as input gives good quality.
Selecting a higher sample rate (96 kHz), which would otherwise give choppy
audio, instead results in an error.

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

Review URL: https://webrtc-codereview.appspot.com/1628004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4198 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-07 17:56:50 +00:00
pwestin@webrtc.org
1064cf06b0 Fixed Rtp/Rtcp tests 
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1627005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4196 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-07 16:03:19 +00:00
andrew@webrtc.org
6367fe885a Fix relative path to .gitignore and other minor changes. 
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1624005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4195 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-07 15:43:04 +00:00
mflodman@webrtc.org
3ba883f0fc Removing functionality for inserting pre-encoded frames instead of raw 
video frames. The functionality hasn't been used for a long time and
should be done properly if used in the future.

This is a pre-step for implementing CPU overload control.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1630004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4194 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-07 13:57:57 +00:00
andrew@webrtc.org
b69cc15467 Add script for appending entries to .gitignore. 
TBR=kjellander

Review URL: https://webrtc-codereview.appspot.com/1629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4193 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-07 02:04:12 +00:00
andrew@webrtc.org
da710448b2 Fix size_t to int conversion error on Win64. 
TBR=pwestin

Review URL: https://webrtc-codereview.appspot.com/1626005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4192 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-07 01:43:12 +00:00
sergeyu@chromium.org
7e4ff354e3 Remove fake screen capturer because it's not used anywhere. 
R=alexeypa@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1625004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4191 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-06 23:11:33 +00:00
pwestin@webrtc.org
8d80fa83fc Fix for STL vector function data not available. 
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1626004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4190 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-06 21:33:06 +00:00
pwestin@webrtc.org
d30859e58e Connect ACM with RTP module for audio NACK. 
Depends on http://review.webrtc.org/1507004/

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1613007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4189 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-06 21:09:01 +00:00
turaj@webrtc.org
a305e9612a Nack for audio. 
R=stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1507004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4188 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-06 19:00:09 +00:00
sergeyu@chromium.org
d9c4658756 Fix leaks in DesktopRegion 
BUG=crbug.com/246870
R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1615004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4186 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-05 19:24:42 +00:00
fischman@webrtc.org
2b3a29a1fa Implement DetectNumberOfCores on Android and make it consistent on Linux and Android 
TEST=Number of cpu cores on Linux and Android is right
R=fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1585007

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4185 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-05 16:37:42 +00:00
pwestin@webrtc.org
db24995680 Wire up Nack for Voe 
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1614004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4184 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-05 15:33:20 +00:00
pbos@webrtc.org
7f1b0ae888 Fix init list for VideoSendStream::Config::Rtp. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1616004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4183 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-05 11:39:18 +00:00
pbos@webrtc.org
025f4f152b Stats+Config moved into VideoSend/ReceiveStreams. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1561006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4182 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-05 11:33:21 +00:00
kjellander@webrtc.org
fec34d7afa Merge webrtc_utility_unittests into modules_unittests. 
This CL eliminates the webrtc_utility_unittests test target.

NOTICE: Upon committing, this test must be removed from the
Buildbot configuration.

BUG=1843
TEST=trybots passing. Compiled and ran modules_unittests, verified the
AudioFrameOperationsTest test executes and passes.

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1584004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4181 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-05 08:58:46 +00:00
andrew@webrtc.org
b2d29bd2fe Restore relative include paths to libyuv. 
Required in order to use an externally compiled libyuv. Removed
in https://code.google.com/p/webrtc/source/detail?r=4167

TBR=tnakamura

Review URL: https://webrtc-codereview.appspot.com/1611005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4180 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 23:53:48 +00:00
turaj@webrtc.org
3942f3a985 Issue 1847, memcopy is wrong and unnecessary, it is sufficient to store the pointer before clearing the instance, and write back the pointer. 
bug=issue1847

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1585006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4178 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 21:31:22 +00:00
fbarchard@google.com
16d78bd307 Fix scale.cc build error with mingw64 -m32 gcc 
BUG=571
TESTED=gcc scale.cc
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1613005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4177 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 19:41:00 +00:00
turaj@webrtc.org
9238de9d49 resolve b9050210. Avoid pushing sync packet before any packet received. Do not turn on AV-sync if initial delay is zero. 
Also solve DTMF playout with Opus. 

issue=b9050210
Test=Manual by QA Team.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1583004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4176 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 19:18:39 +00:00
sergeyu@chromium.org
3d34f66292 Move screen capturers from chromium to webrtc. 
R=alexeypa@chromium.org, wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1586005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4175 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 18:51:23 +00:00
fischman@webrtc.org
b7a8f43670 Roll chromium_revision in webrtc 199267:203806 
This switches the default build system on linux from make to ninja.  Details in
https://groups.google.com/a/chromium.org/forum/?fromgroups#!topic/chromium-dev/_Fsv4_XZ_bo

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1607004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4174 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 17:10:24 +00:00
kjellander@webrtc.org
430464c776 Add WebKit/Tools/Scripts to support Android test execution. 
In https://code.google.com/p/webrtc/source/detail?r=4038 we rolled
chromium_revision past the point where WebKit/Tools/Scripts had its
own DEP in the Chromium DEPS file.
Since Chromium now only have a single WebKit checkout, we need to
pull the Tools/Scripts dir to be able to use the Android test
framework (build/android/run_test.py) since it's depending on modules
in webkitpy.

I have filed http://crbug.com/246529 to get this dependency removed.

BUG=1882
TEST=build/android/run_tests.py executes without any import errors.
R=fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1608004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4173 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 16:29:45 +00:00
stefan@webrtc.org
a817962bab Refactor padding and rtp header functionality. 
BUG=1837
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1611004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4172 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 13:47:36 +00:00
stefan@webrtc.org
de98478965 Update the remote bitrate estimator before passing the packet to the RTP module. 
This solves the problem of reconstructed packets biasing the bandwidth estimate.

TEST=vie_auto_test --automated, trybots
R=mflodman@webrtc.org, solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1594005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4171 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 12:15:40 +00:00
pbos@webrtc.org
6998c8ef7a Remove XvRenderer. 
One test renderer per platform is sufficient, multiple code paths are
bad.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1612004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4170 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 11:56:06 +00:00
stefan@webrtc.org
8ad3ec9722 Fix build error introduced with r4168. 
TBR=mflodman@webrtc.org
BUG=1837

Review URL: https://webrtc-codereview.appspot.com/1610004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4169 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 09:52:46 +00:00
stefan@webrtc.org
c3cc375499 Add support for padding in pacer. 
This improves pacer-based padding by making sure it limits padding according to:
- Never pad more than 800 kbps.
- Padding + media should not go above a given target bitrate.

Also adds appropriate unittests to make sure we reach the given targets.

BUG=1837
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1582005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4168 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 09:36:56 +00:00
pbos@webrtc.org
c69ae69d0b Include files from webrtc/.. paths in common_video/ 
BUG=1662
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1546004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4167 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 09:02:37 +00:00
pbos@webrtc.org
ba7f6a8614 Include files from webrtc/.. paths in tools/ 
BUG=1662
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1547004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4166 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 08:14:10 +00:00
kjellander@webrtc.org
5156c94f89 Disable neteq_unittests on Win x64 in code. 
Having this failing test being disabled in code will make it
possible to add it on the bots again, and make thus no bot
configuration update needs to be communicated when it's fixed.

BUG=1460
TEST=Compiled with GYP_DEFINES=target_arch=x64 and ran the
test successsfully on Windows. Also ran regular trybots.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1595004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4165 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 05:47:24 +00:00
kjellander@webrtc.org
b6e49aa3f2 Disable audio_decoder_unittests on Win x64 in code. 
Having this failing test being disabled in code will make it
possible to add it on the bots again, and make thus no bot
configuration update needs to be communicated when it's fixed.

BUG=1459
TEST=Compiled with GYP_DEFINES=target_arch=x64 and ran the
test successsfully on Windows. Also ran regular trybots.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1594004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4164 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 05:47:04 +00:00
kjellander@webrtc.org
6eba2774c9 Disable audio_coding_unittests on Win x64 in code. 
Having this failing test being disabled in code will make it
possible to add it on the bots again, and make thus no bot
configuration update needs to be communicated when it's fixed.

BUG=1458
TEST=Compiled with GYP_DEFINES=target_arch=x64 and ran the
test successsfully on Windows. Also ran regular trybots.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1593004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4163 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 05:46:37 +00:00
fischman@webrtc.org
e001b57d84 Do not hold a lock when calling VCMReceiveCallback::FrameToRender. 
DecodedImageCallback is allowed to be called on a thread different from decoding thread. To avoid the deadlock in VCMDecodedFrameCallback::Decoded, VCMDecodedFrameCallback::_critSect  should not be held while calling VCMReceiveCallback::FrameToRender.

BUG=1832
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1570004

Patch from Wu-Cheng Li <wuchengli@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4162 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 03:29:37 +00:00
sergeyu@chromium.org
3ee13e4ac2 Optimized DesktopRegion implementation. 
Now DestktopRegion can merge overlapping rectangles.

R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1526004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4161 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 00:38:39 +00:00
fischman@webrtc.org
34a77354a8 Removed unused class members to enable clang=1 android build. 
BUG=https://code.google.com/p/webrtc/issues/detail?id=1275
TESTED=video_demo_apk builds with clang=1
R=niklas.enbom@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1605004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4160 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 00:37:21 +00:00
mikhal@webrtc.org
6eb0f6a4d9 Setting SSRC in vie_loopback_test 
BUG=1822
R=pwestin@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1603004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4159 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-03 22:54:40 +00:00
andrew@webrtc.org
0a38432ea5 Fix error in mixing test for supported sample rates. 
With the switch to an arbitrary resampler, we now support these strange
rates.

TBR=turaj

Review URL: https://webrtc-codereview.appspot.com/1604004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4158 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-03 22:52:09 +00:00
wu@webrtc.org
fa64a595ad Change SetRTPAudioLevelIndicationStatus to ignore the id in the case of disabling. 
This makes it easier for the users of the interface, i.e. doesn't need to remember the id in order to disable audio level indication later.

BUG=1828
TEST=unit tests
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1598005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4157 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-03 21:27:57 +00:00
andrew@webrtc.org
c1eb560a5c Replace the old resampler with SincResampler in the voice engine signal path. 
* The old resampler was found to have a wraparound bug.
* Remove support for the old resampler from PushResampler.
* Use PushResampler in AudioCodingModule.
* The old resampler must still be removed from the file utility.

BUG=webrtc:1867,webrtc:827
TESTED=unit tests, Chrome using apprtc and voe_cmd_test to verify wrap-around is corrected, voe_cmd_test running through all supported codec sample rates and channels to verify good quality audio
R=henrika@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1590004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4156 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-03 19:00:29 +00:00
andrew@webrtc.org
31c5f1c91a Remove ancient and unused CNG test. 
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1585005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4154 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-03 16:07:07 +00:00