external/webrtc.git
2 min ago(Auto)update libjingle 75610402-> 75610402 master
buildbot@webrtc.org [Tue, 16 Sep 2014 15:24:15 +0000 (15:24 +0000)]
(Auto)update libjingle 75610402-> 75610402

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7194 4adac7df-926f-26a2-2b94-8c16560cd09d

4 hours agortc_unittest: prevent execution of broken tests.
henrike@webrtc.org [Tue, 16 Sep 2014 11:19:32 +0000 (11:19 +0000)]
rtc_unittest: prevent execution of broken tests.

BUG=1976
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7193 4adac7df-926f-26a2-2b94-8c16560cd09d

4 hours agoFix GN for rtc_base_approved target.
kjellander@webrtc.org [Tue, 16 Sep 2014 11:16:12 +0000 (11:16 +0000)]
Fix GN for rtc_base_approved target.

In https://webrtc-codereview.appspot.com/22649004
a new target was introduced that duplicated some
source files, breaking the bots in
http://build.chromium.org/p/chromium.webrtc.fyi/waterfall

This updates the GN config to also remove them from
the target where they were moved from in base.gyp.

BUG=3806
TESTED=Trybots + Running GN in a Chromium checkout with
src/third_party/webrtc symlinked to the WebRTC checkout
with this CL applied + passing compile step.

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7192 4adac7df-926f-26a2-2b94-8c16560cd09d

5 hours agomemcheck: suppressions didn't map over directly when moving base from talk to webrtc...
henrike@webrtc.org [Tue, 16 Sep 2014 09:41:21 +0000 (09:41 +0000)]
memcheck: suppressions didn't map over directly when moving base from talk to webrtc (part of the suppression that is not related to the signature differed). Fixed suppressions accordingly.

BUG=N/A
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7191 4adac7df-926f-26a2-2b94-8c16560cd09d

6 hours agoRevert 7184 "Enable ipv6 by default for webrtc under a Finch exp..."
kjellander@webrtc.org [Tue, 16 Sep 2014 08:58:22 +0000 (08:58 +0000)]
Revert 7184 "Enable ipv6 by default for webrtc under a Finch exp..."

Breaks Chrome build and prevents rolling WebRTC into Chrome DEPS.

> Enable ipv6 by default for webrtc under a Finch experiment.
>
> BUG=413437 (chromium)
> https://code.google.com/p/chromium/issues/detail?id=413437
>
> Review URL: https://webrtc-codereview.appspot.com/23529005

TBR=guoweis@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7190 4adac7df-926f-26a2-2b94-8c16560cd09d

10 hours agoaudio_processing/aec: Ported NEON optimizations of SubbandCoherence() and its sub...
bjornv@webrtc.org [Tue, 16 Sep 2014 05:01:42 +0000 (05:01 +0000)]
audio_processing/aec: Ported NEON optimizations of SubbandCoherence() and its sub-functions to SSE2

These optimizations were originally committed in r6860, but reverted in r6861, since it broke a bitexactness test (ApmTest.Process) in modules_unittests. That test has now been updated in r7149, hence this CL now pass the test.

BUG=3767
TESTED=manually on linux and trybots
TBR=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7189 4adac7df-926f-26a2-2b94-8c16560cd09d

14 hours agoAdd a target for the approved subset of rtc_base.
andrew@webrtc.org [Tue, 16 Sep 2014 01:03:29 +0000 (01:03 +0000)]
Add a target for the approved subset of rtc_base.

rtc_base drags in a bunch of unwieldly dependencies (e.g. nss and
json) not required for standalone webrtc (aka rtc/media). The root of
the problem appears to be that MessageQueue depends on a socket server.
(And since common.h -> logging.h -> thread.h -> messagequeue.h, this
dependency spreads quickly.)

This starts a new target for a "purified" subset of rtc_base. It adds
the files which are already being used, replacing the use of common.h
with checks.h. desktop_capture is a lost cause, and retains its
dependency on the full rtc_base.

The hope is that as additional components are desired they will be
cleaned and added to rtc_base_approved.

BUG=3806
R=andresp@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7188 4adac7df-926f-26a2-2b94-8c16560cd09d

19 hours agoFix memory leak in webrtc::MouseCursorMonitorMac
sergeyu@chromium.org [Mon, 15 Sep 2014 20:11:23 +0000 (20:11 +0000)]
Fix memory leak in webrtc::MouseCursorMonitorMac

BUG=3815
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/24579004

Patch from Vicken Simonian <vsimon@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7187 4adac7df-926f-26a2-2b94-8c16560cd09d

20 hours agoPartial implementation of rtc::LogMessage in chromium overrides.
glaznev@webrtc.org [Mon, 15 Sep 2014 19:16:21 +0000 (19:16 +0000)]
Partial implementation of rtc::LogMessage in chromium overrides.

rtc::LogMessage::LogToDebug used in peerconnection_jni.cc.

BUG=https://crbug.com/412276
R=glaznev@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7186 4adac7df-926f-26a2-2b94-8c16560cd09d

21 hours agoHW video decoding optimization to better support HD resolution:
glaznev@webrtc.org [Mon, 15 Sep 2014 17:52:42 +0000 (17:52 +0000)]
HW video decoding optimization to better support HD resolution:

- Change hw video decoder wrapper to allow to feed multiple input
and query for an output every 10 ms.
- Add an option to decode video frame into an Android surface object. Create
shared with video renderer EGL context and external texture on
video decoder thread.
- Support external texture rendering in Android renderer.
- Support TextureVideoFrame in Java and use it to pass texture from video decoder
to renderer.
- Fix HW encoder and decoder detection code to avoid query codec capabilities
from sw codecs.

BUG=
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7185 4adac7df-926f-26a2-2b94-8c16560cd09d

22 hours agoEnable ipv6 by default for webrtc under a Finch experiment.
guoweis@webrtc.org [Mon, 15 Sep 2014 16:31:13 +0000 (16:31 +0000)]
Enable ipv6 by default for webrtc under a Finch experiment.

BUG=413437 (chromium)
https://code.google.com/p/chromium/issues/detail?id=413437

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7184 4adac7df-926f-26a2-2b94-8c16560cd09d

24 hours agoMake BW checks > 0 in peerconnection_unittest.cc.
pbos@webrtc.org [Mon, 15 Sep 2014 14:38:07 +0000 (14:38 +0000)]
Make BW checks > 0 in peerconnection_unittest.cc.

These checks (> 40k) fail on LSan FYI bots and the purpose of them seem
to be that we're getting non-zero BW reported.

R=stefan@webrtc.org
TBR=jiayl@webrtc.org, solenberg@webrtc.org
BUG=3817,chromium:375154

Review URL: https://webrtc-codereview.appspot.com/29479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7183 4adac7df-926f-26a2-2b94-8c16560cd09d

26 hours agoaudio_processing: Correct sample rate in aec_debug_dump
bjornv@webrtc.org [Mon, 15 Sep 2014 13:23:07 +0000 (13:23 +0000)]
audio_processing: Correct sample rate in aec_debug_dump

When writing to wav files in the low level flag aec_debug_dump incorrect sample rates were used for recordings using rates from 32 kHz and above. This since internally inside the AEC we process the data using 16 kHz. Any upper band is processed and combined later on.

This CL adds the correct sample rate to the recording.

BUG=3359
TESTED=locally on 44.1 kHz recordings on Linux
R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7182 4adac7df-926f-26a2-2b94-8c16560cd09d

26 hours agoRe-enable neteq_performance_unittest.cc for android.
andresp@webrtc.org [Mon, 15 Sep 2014 12:29:50 +0000 (12:29 +0000)]
Re-enable neteq_performance_unittest.cc for android.

BUG=3770
R=kjellander@webrtc.org
TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7181 4adac7df-926f-26a2-2b94-8c16560cd09d

26 hours agoRe-enable rampup_tests.cc for Android.
andresp@webrtc.org [Mon, 15 Sep 2014 12:27:35 +0000 (12:27 +0000)]
Re-enable rampup_tests.cc for Android.

BUG=3770
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7180 4adac7df-926f-26a2-2b94-8c16560cd09d

27 hours agoRe-enable video send stream tests for android.
andresp@webrtc.org [Mon, 15 Sep 2014 12:24:34 +0000 (12:24 +0000)]
Re-enable video send stream tests for android.

BUG=3770
R=kjellander@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7179 4adac7df-926f-26a2-2b94-8c16560cd09d

28 hours agoFix ThreadChecker unittests when DCHECK_ALWAYS_ON is defined
henrik.lundin@webrtc.org [Mon, 15 Sep 2014 11:19:35 +0000 (11:19 +0000)]
Fix ThreadChecker unittests when DCHECK_ALWAYS_ON is defined

This requires two fixes:
1. Use DCHECK instead of assert in ThreadChecker's unittest.

2. Activate DCHECK when DCHECK_ALWAYS_ON in enabled.

Both these modifications are in line with Chromium's implementation.
The ThreadChecker unittest was changed to use assert instead of DCHECK
on the initial import (since WebRTC did not have a DCHECK back then).

BUG=3803
TEST=local out/{Debug,Release}/rtc_unittests built with and without DCHECK_ALWAYS_ON
R=andrew@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7178 4adac7df-926f-26a2-2b94-8c16560cd09d

31 hours agoStop building talk/xmllite since it is no longer used.
henrike@webrtc.org [Mon, 15 Sep 2014 08:13:36 +0000 (08:13 +0000)]
Stop building talk/xmllite since it is no longer used.

BUG=3379
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7176 4adac7df-926f-26a2-2b94-8c16560cd09d

31 hours agoSet minimum SDK level to 10.7 for Mac and iOS.
kjellander@webrtc.org [Mon, 15 Sep 2014 08:02:43 +0000 (08:02 +0000)]
Set minimum SDK level to 10.7 for Mac and iOS.

This is needed since r7174 introduced a dependency
on AVFoundation, which is not present in the 10.6 SDK which is
still the default for Chromium.

BUG=
TESTED=Passing compile on trybots.
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7175 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days ago(Auto)update libjingle 75390072-> 75428737
buildbot@webrtc.org [Sat, 13 Sep 2014 01:09:18 +0000 (01:09 +0000)]
(Auto)update libjingle 75390072-> 75428737

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7174 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoRevert 7170 "Revert 7121 "ValidateFrame, When dumping the first ..."
fbarchard@google.com [Sat, 13 Sep 2014 00:52:42 +0000 (00:52 +0000)]
Revert 7170 "Revert 7121 "ValidateFrame, When dumping the first ..."
BUG=3789
TESTED=drmemory out\Debug\libjingle_media_unittest.exe --gtest_catch_exceptions=0 --gtest_filter=*Validate*

> Revert 7121 "ValidateFrame, When dumping the first 4 samples of a frame, first copy it to a temporary buffer that is zero padded, them use that."
>
> Breaks other repos.
>
> TBR=fbarchard@google.com
> BUG=N/A
>
> Review URL: https://webrtc-codereview.appspot.com/23639004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7173 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoAdd enable flag for Android device orientation change event.
glaznev@webrtc.org [Fri, 12 Sep 2014 16:48:12 +0000 (16:48 +0000)]
Add enable flag for Android device orientation change event.

There are reports (not reproducible with appRtcDemo) that
outstanding device orientation change event
OrientationEventListener.onOrientationChanged can be
triggered even after these events are disabled by
OrientationEventListener.disable() code.
Avoid calling native code in this case since underlying
C++ class may have already been deleted.

BUG=3564
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7172 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoTemporary revert maximum video codec resolution back to 1080p.
glaznev@webrtc.org [Fri, 12 Sep 2014 16:40:35 +0000 (16:40 +0000)]
Temporary revert maximum video codec resolution back to 1080p.

BUG=3757, 3738
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7171 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoRevert 7121 "ValidateFrame, When dumping the first 4 samples of a frame, first copy...
henrike@webrtc.org [Fri, 12 Sep 2014 16:31:29 +0000 (16:31 +0000)]
Revert 7121 "ValidateFrame, When dumping the first 4 samples of a frame, first copy it to a temporary buffer that is zero padded, them use that."

Breaks other repos.

TBR=fbarchard@google.com
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/23639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7170 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoInitialize restored_packet in nack_rtx_unittest.cc.
pbos@webrtc.org [Fri, 12 Sep 2014 16:16:00 +0000 (16:16 +0000)]
Initialize restored_packet in nack_rtx_unittest.cc.

This is to get the DrMemory Full bots to go green, this was previously
suppressed. This fix is likely hiding a real bug that should be
investigated, but it's not a regression from before. The issue should
not be closed before we figure out why this is the case and revert this
"fix".

TBR=stefan@webrtc.org
BUG=3183

Review URL: https://webrtc-codereview.appspot.com/30369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7169 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agolinux: remove stray libcrypto dependency
henrike@webrtc.org [Fri, 12 Sep 2014 16:11:38 +0000 (16:11 +0000)]
linux: remove stray libcrypto dependency

Followup to CL 20049004, which looks like it added an unneeded -lcrypto
on linux.

BUG=3625
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7168 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoDisable MethodNotAllowedOnDifferentThreadInDebug.
henrike@webrtc.org [Fri, 12 Sep 2014 15:57:08 +0000 (15:57 +0000)]
Disable MethodNotAllowedOnDifferentThreadInDebug.

BUG=3803
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7167 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoFix mac video_render implementation on cocoa.
andresp@webrtc.org [Fri, 12 Sep 2014 13:57:47 +0000 (13:57 +0000)]
Fix mac video_render implementation on cocoa.

Hit this while playing around with all compile possibilities for issue 3770.

BUG=3770
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7166 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoFix stack limit exceeded in http client.
andresp@webrtc.org [Fri, 12 Sep 2014 13:35:05 +0000 (13:35 +0000)]
Fix stack limit exceeded in http client.

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7165 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoAdd ability to downscale content to improve quality.
pbos@webrtc.org [Fri, 12 Sep 2014 11:51:47 +0000 (11:51 +0000)]
Add ability to downscale content to improve quality.

BUG=3712
R=marpan@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7164 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoMake RTPSender/RTPReceiver generic.
pbos@webrtc.org [Fri, 12 Sep 2014 11:05:55 +0000 (11:05 +0000)]
Make RTPSender/RTPReceiver generic.

Changes include,
1) Introduce class RtpPacketizerGeneric & RtpDePacketizerGeneric.
2) Introduce class RtpDepacketizerVp8.
3) Make RTPSenderVideo::SendH264 generic and used by all packetizers.
4) Move codec specific functions from RTPSenderVideo/RTPReceiverVideo to
RtpPacketizer/RtpDePacketizer sub-classes.

R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26399004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7163 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoMark all virtual overrides in the hierarchy of RtpData and RtpReceiver as such.
stefan@webrtc.org [Fri, 12 Sep 2014 07:42:33 +0000 (07:42 +0000)]
Mark all virtual overrides in the hierarchy of RtpData and RtpReceiver as such.

This will make further changes to these classes safer by ensuring that the
compile breaks if the base class changes and not all overrides are fixed.

This also highlighted a number of unused functions which I've removed.

-- This is was reviewed in https://webrtc-codereview.appspot.com/19309004/, but
-- a new cl was needed to resolve a small conflict before committing.

BUG=none
TEST=none
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7162 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoMark all virtual overrides in the hierarchies of RtpDump and
henrike@webrtc.org [Thu, 11 Sep 2014 22:45:54 +0000 (22:45 +0000)]
Mark all virtual overrides in the hierarchies of RtpDump and
VCMPacketizationCallback as such.

This will make further changes to these classes safer by ensuring that the
compile breaks if the base class changes and not all overrides are fixed.

This also marks all other such overrides in the affected files.

BUG=none
TEST=none
R=henrike@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7161 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days ago(Auto)update libjingle 75302540-> 75327856
buildbot@webrtc.org [Thu, 11 Sep 2014 21:52:48 +0000 (21:52 +0000)]
(Auto)update libjingle 75302540-> 75327856

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7160 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoUpdate AUTHORS file.
henrike@webrtc.org [Thu, 11 Sep 2014 21:12:59 +0000 (21:12 +0000)]
Update AUTHORS file.

BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7159 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoFix window capturing on Windows when the window is minimized.
jiayl@webrtc.org [Thu, 11 Sep 2014 19:33:58 +0000 (19:33 +0000)]
Fix window capturing on Windows when the window is minimized.

BUG=crbug/410290
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/20319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7158 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoSkip dlclose() on AddressSanitizer.
pbos@webrtc.org [Thu, 11 Sep 2014 17:29:11 +0000 (17:29 +0000)]
Skip dlclose() on AddressSanitizer.

AddressSanitizer can't symbolize parts of the stack that contains
dlclose()d modules. This makes some LSan suppressions not kick in and
blocks launching the LSan bot for WebRTC.
This "fix" excludes dlclose() in
webrtc/modules/audio_device/linux/latebindingsymboltable_linux.cc which
resolves this on the bot.

R=xians@webrtc.org
BUG=3402,chromium:375154

Review URL: https://webrtc-codereview.appspot.com/25499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7157 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoStop building talk/sound since it is no longer used.
henrike@webrtc.org [Thu, 11 Sep 2014 17:16:56 +0000 (17:16 +0000)]
Stop building talk/sound since it is no longer used.

BUG=N/A
R=pbos@webrtc.org
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7156 4adac7df-926f-26a2-2b94-8c16560cd09d

4 days agoDisabling initializeAndroidGlobals when built with WEBRTC_CHROMIUM_BUILD.
glaznev@webrtc.org [Thu, 11 Sep 2014 16:58:25 +0000 (16:58 +0000)]
Disabling initializeAndroidGlobals when built with WEBRTC_CHROMIUM_BUILD.

webrtc::VideoEngine::SetAndroidObjects and webrtc::VoiceEngine::SetAndroidObjects
are not compatible with WEBRTC_CHROMIUM_BUILD. Since neither VoiceEngine nor VideoEngine
are needed at the time it's better to disable it completely.

BUG=https://crbug.com/412276
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7155 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoSplit suppressons of thread.cc and messagequeue.cc.
pbos@webrtc.org [Thu, 11 Sep 2014 14:59:06 +0000 (14:59 +0000)]
Split suppressons of thread.cc and messagequeue.cc.

Most calls have either of these in the stack, meaning that pretty much
all races are suppressed.

BUG=
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7154 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoRemove developing code in ns_core
aluebs@webrtc.org [Thu, 11 Sep 2014 11:19:56 +0000 (11:19 +0000)]
Remove developing code in ns_core

This defines were hardcoded and the code inside of the ifdefs was never used.

BUG=webrtc:3763
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7153 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoAdd myself to common_audio and audio_processing watchlists
aluebs@webrtc.org [Thu, 11 Sep 2014 10:11:43 +0000 (10:11 +0000)]
Add myself to common_audio and audio_processing watchlists

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7152 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoRevert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."
henrikg@webrtc.org [Thu, 11 Sep 2014 09:48:30 +0000 (09:48 +0000)]
Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."

Speculative revert, seems to be reason for flaky Win FYI bot compile break.

> Expose VideoEncoders with webrtc/video_encoder.h.
>
> Exposes VideoEncoders as part of the public API and provides a factory
> method for creating them.
>
> BUG=3070
> R=mflodman@webrtc.org, stefan@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/21929004

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7151 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoRestore webrtc_base target until r7140 is rolled into Chromium.
kjellander@webrtc.org [Thu, 11 Sep 2014 09:22:13 +0000 (09:22 +0000)]
Restore webrtc_base target until r7140 is rolled into Chromium.

In r7140 the webrtc_base target was renamed to rtc_base. This
breaks our FYI bots for rolling WebRTC in Chromium's DEPS.
By re-adding a None target named webrtc_base, this transition
should be smoother.

TBR=henrikg@webrtc.org,
TESTED=Passed build/gyp_chromium on a Chromium checkout with src/third_party/webrtc replaced by a mount like this:
cd /path/to/chromium/src
sudo mount --bind /path/to/webrtc/trunk/webrtc third_party/webrtc

Review URL: https://webrtc-codereview.appspot.com/23589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7150 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoaudio_processing_unittests: Enabled ApmTest.Process for all platforms but Android
bjornv@webrtc.org [Thu, 11 Sep 2014 08:36:35 +0000 (08:36 +0000)]
audio_processing_unittests: Enabled ApmTest.Process for all platforms but Android

During porting some neon optimizations to sse2 the ApmTest.Process failed despite bit exact outputs. The reason is that with float data between component, as to previously truncating to int, we get small deviations in logged metrics. This affected a noise probability to a small fraction, which is not a particular bug.

This CL change the comparison from EXPECT_EQ() to EXPECT_NEAR() which then as a result makes the test run on Mac and Windows as well.

For int values a deviation of 1 is acceptable, which would include any rounding errors.
For float values a deviation of 0.0005 is chosen by looking at current test stats for the affected platforms/optimizations.

BUG=114
TESTED=locally on linux with and without sse2 optimizations and trybots
R=aluebs@webrtc.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7149 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoRevert 7145 "Stop building talk/sound since it is no longer used."
sprang@webrtc.org [Thu, 11 Sep 2014 08:29:53 +0000 (08:29 +0000)]
Revert 7145 "Stop building talk/sound since it is no longer used."

> Stop building talk/sound since it is no longer used.
>
> BUG=N/A
> R=pbos@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/22319004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7148 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoCalculating round-trip-time in send-only channel in VoE.
minyue@webrtc.org [Thu, 11 Sep 2014 07:51:53 +0000 (07:51 +0000)]
Calculating round-trip-time in send-only channel in VoE.

TESTS=built chromium and tested with 1:1 hangout call

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

Review URL: https://webrtc-codereview.appspot.com/23489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7147 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoMark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE.
henrik.lundin@webrtc.org [Thu, 11 Sep 2014 06:20:28 +0000 (06:20 +0000)]
Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE.

This will make a subsequent change I intend to do safer, where I'll change the
return type of one of the base Module functions, by breaking the compile if I
miss any overrides.

This also highlighted a number of unused functions (in many cases apparently
virtual "overrides" of no-longer-existent base functions).  I've removed some of
these.

This also highlighted several cases where "virtual" was used unnecessarily to
mark a function that was only defined in one class.  Removed "virtual" in those
cases.

BUG=none
TEST=none
R=andrew@webrtc.org, henrik.lundin@webrtc.org, mallinath@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7146 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoStop building talk/sound since it is no longer used.
henrike@webrtc.org [Wed, 10 Sep 2014 22:18:04 +0000 (22:18 +0000)]
Stop building talk/sound since it is no longer used.

BUG=N/A
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7145 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoMark all virtual overrides in the hierarchy of AudioPacketizationCallback,
henrike@webrtc.org [Wed, 10 Sep 2014 22:14:59 +0000 (22:14 +0000)]
Mark all virtual overrides in the hierarchy of AudioPacketizationCallback,
RTPStream, and NetEq as such.  Also mark all other virtual overrides in the same
files.

This will make further changes to these classes safer by ensuring that the
compile breaks if the base class changes and not all overrides are fixed.

This also deletes ACMTest.cc, which existed solely to define ~ACMTest(), which
was marked pure virtual in the header.  (Pure virtual destructors still need a
definition.)  Because there is another pure virtual method in this class, the
class is already abstract, so there's no benefit to making the desturctor pure.
Making it non-pure allows removing the separate source file.

BUG=none
TEST=none
R=henrik.lundin@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7144 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoFix MSVC warnings about value truncations, webrtc/base/ edition.
henrike@webrtc.org [Wed, 10 Sep 2014 22:10:24 +0000 (22:10 +0000)]
Fix MSVC warnings about value truncations, webrtc/base/ edition.

BUG=chromium:81439
TEST=none
R=henrike@webrtc.org, marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/20249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7143 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoFix frame rate selection for Android camera.
glaznev@webrtc.org [Wed, 10 Sep 2014 19:24:57 +0000 (19:24 +0000)]
Fix frame rate selection for Android camera.

- Android camera supports multiple fps values for a single video
resolution - change video source default video format selection
to pick up best available fps.
- Change fps range calculation to better match target fps value.

BUG=2622
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7142 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoAdd schannel webrtc_base build using a new use_schannel gyp variable.
tpsiaki@google.com [Wed, 10 Sep 2014 18:06:47 +0000 (18:06 +0000)]
Add schannel webrtc_base build using a new use_schannel gyp variable.

R=henrike@webrtc.org, thorcarpenter@google.com

Review URL: https://webrtc-codereview.appspot.com/28409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7141 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoPut base tests in webrtc_tests.gyp
henrike@webrtc.org [Wed, 10 Sep 2014 17:28:19 +0000 (17:28 +0000)]
Put base tests in webrtc_tests.gyp

BUG=N/A
R=andrew@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7140 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoRoll chromium_revision ea769fd..6455c69 (re-land)
kjellander@webrtc.org [Wed, 10 Sep 2014 16:51:37 +0000 (16:51 +0000)]
Roll chromium_revision ea769fd..6455c69 (re-land)

Mainly to pick up https://codereview.chromium.org/552013004

Summary of changes (git diff ea769fd..6455c69 DEPS):
* third_party/libvpx ceebbcc0..d95585f
* third_party/swarming e7d8b98..14b5fd82
* tools/gyp 1972:1973

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7139 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoEnable shared socket for TurnPort.
jiayl@webrtc.org [Wed, 10 Sep 2014 16:31:34 +0000 (16:31 +0000)]
Enable shared socket for TurnPort.
In AllocationSequence::OnReadPacket, we now hand the packet to both the TurnPort and StunPort if the remote address matches the server address.

TESTED=AppRtc loopback call generates both turn and stun candidates.

BUG=1746
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7138 4adac7df-926f-26a2-2b94-8c16560cd09d

5 days agoConvert GN visibility to be lists.
brettw@chromium.org [Wed, 10 Sep 2014 16:24:11 +0000 (16:24 +0000)]
Convert GN visibility to be lists.

This is a followup to my previous patch that missed this case.

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7137 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoRemove suppressions for VideoFrame::Validate.
pbos@webrtc.org [Wed, 10 Sep 2014 14:59:09 +0000 (14:59 +0000)]
Remove suppressions for VideoFrame::Validate.

BUG=3789
R=sprang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7136 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoSimplify gyp rules on video_render_module.
andresp@webrtc.org [Wed, 10 Sep 2014 14:48:48 +0000 (14:48 +0000)]
Simplify gyp rules on video_render_module.

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7135 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoFix printing of error stack in rtcbot when a test fails via test.fail().
houssainy@google.com [Wed, 10 Sep 2014 14:35:35 +0000 (14:35 +0000)]
Fix printing of error stack in rtcbot when a test fails via test.fail().

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7134 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoFix compile error on JDK 1.7.
kjellander@webrtc.org [Wed, 10 Sep 2014 12:35:59 +0000 (12:35 +0000)]
Fix compile error on JDK 1.7.

JDK 1.7 gives an error like this:
warning: [static] static method should be qualified by type name

R=pbos@webrtc.org
TBR=henrike@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/29399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7133 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoRoll gtest-parallel.
pbos@webrtc.org [Wed, 10 Sep 2014 09:29:12 +0000 (09:29 +0000)]
Roll gtest-parallel.

Brings in change that eliminates Queues which shows significant speed
improvement for huge work lists.

R=andrew@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/26409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7132 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoRemove DestructEncoderInst and its codec-specific implementations.
henrik.lundin@webrtc.org [Wed, 10 Sep 2014 08:52:26 +0000 (08:52 +0000)]
Remove DestructEncoderInst and its codec-specific implementations.

This method is seemingly never called.

BUG=none
TEST=none
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7131 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoRevert 7128 "Roll chromium_revision ea769fd..6455c69"
kjellander@webrtc.org [Wed, 10 Sep 2014 08:38:27 +0000 (08:38 +0000)]
Revert 7128 "Roll chromium_revision ea769fd..6455c69"

> Roll chromium_revision ea769fd..6455c69
>
> Mainly to pick up https://codereview.chromium.org/552013004
>
> Summary of changes (git diff ea769fd..6455c69 DEPS):
> * third_party/libvpx ceebbcc0..d95585f
> * third_party/swarming e7d8b98..14b5fd82
> * tools/gyp 1972:1973
>
> R=pbos@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/29379004

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7130 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days ago(Auto)update libjingle 75141932-> 75179475
buildbot@webrtc.org [Wed, 10 Sep 2014 07:57:12 +0000 (07:57 +0000)]
(Auto)update libjingle 75141932-> 75179475

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7129 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoRoll chromium_revision ea769fd..6455c69
kjellander@webrtc.org [Wed, 10 Sep 2014 07:42:55 +0000 (07:42 +0000)]
Roll chromium_revision ea769fd..6455c69

Mainly to pick up https://codereview.chromium.org/552013004

Summary of changes (git diff ea769fd..6455c69 DEPS):
* third_party/libvpx ceebbcc0..d95585f
* third_party/swarming e7d8b98..14b5fd82
* tools/gyp 1972:1973

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7128 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoinclude cstdlib for free() and abort()
andrew@webrtc.org [Wed, 10 Sep 2014 03:24:36 +0000 (03:24 +0000)]
include cstdlib for free() and abort()

This previous CL added uses of free() and abort() without including cstdlib:
https://webrtc-codereview.appspot.com/22449004

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23559004

Patch from Mostyn Bramley-Moore <mostynb@opera.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7127 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoAdd a new class InterfaceAddress inherited from IPAddress to keep track of IPv6 Addre...
guoweis@webrtc.org [Tue, 9 Sep 2014 23:42:40 +0000 (23:42 +0000)]
Add a new class InterfaceAddress inherited from IPAddress to keep track of IPv6 Address flags.

Skeleton put in place in Network::GetFilterIPs() which will be used to
filter addresses

BUG=3773
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7126 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoFix up configs applying to GN build.
brettw@chromium.org [Tue, 9 Sep 2014 23:34:56 +0000 (23:34 +0000)]
Fix up configs applying to GN build.

The audio_processing target didn't have the build configs applying to it which led to some logging errors.

TBR=kjellander

Review URL: https://webrtc-codereview.appspot.com/22339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7125 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoFixes two issues in how we handle OfferToReceiveX for CreateOffer:
jiayl@webrtc.org [Tue, 9 Sep 2014 21:43:15 +0000 (21:43 +0000)]
Fixes two issues in how we handle OfferToReceiveX for CreateOffer:
1. the options set in the first CreateOffer call should not affect the result of a second CreateOffer call, if SetLocalDescription is not called after the first CreateOffer. So the member var options_ of MediaStreamSignaling is removed to make each CreateOffer independent.
Instead, MediaSession is responsible to make sure that an m-line in the current local description is never removed from the newly created offer.

2. OfferToReceiveAudio used to default to true, i.e. always having m=audio line even if no audio track. This is changed so that by default m=audio is only added if there are any audio tracks.

BUG=2108
R=pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7068

Review URL: https://webrtc-codereview.appspot.com/16309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7124 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoChange explicit static cast from int to uint16_t to implicit cast of 0u.
fbarchard@google.com [Tue, 9 Sep 2014 21:37:27 +0000 (21:37 +0000)]
Change explicit static cast from int to uint16_t to implicit cast of 0u.
BUG=3663
TESTED=local windows build with VS2013.
R=harryjin@google.com, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7123 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoFix the RTC+Chromium GN build.
brettw@chromium.org [Tue, 9 Sep 2014 19:15:33 +0000 (19:15 +0000)]
Fix the RTC+Chromium GN build.

LOGGING_INSIDE_WEBRTC was being set in the inherited config, whereas in the GYP build this define is not inherited. This caused duplicate logging macros to be defined in Chrome files dependening on WebRTC targets.

Move LOGGING_INSIDE_WEBRTC to the common config (non-inherited).

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7122 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoValidateFrame, When dumping the first 4 samples of a frame, first copy it to a tempor...
fbarchard@google.com [Tue, 9 Sep 2014 18:34:53 +0000 (18:34 +0000)]
ValidateFrame, When dumping the first 4 samples of a frame, first copy it to a temporary buffer that is zero padded, them use that.
BUG=3789
TESTED=drmemory out\Debug\libjingle_media_unittest.exe --gtest_catch_exceptions=0 --gtest_filter=*Validate*
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7121 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoTurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MI...
jiayl@webrtc.org [Tue, 9 Sep 2014 15:44:05 +0000 (15:44 +0000)]
TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH.

BUG=3570
R=juberti@webrtc.org, mallinath@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7070

Review URL: https://webrtc-codereview.appspot.com/20999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7120 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoBot Browser files moved to /bot/browser/
houssainy@google.com [Tue, 9 Sep 2014 14:50:09 +0000 (14:50 +0000)]
Bot Browser files moved to /bot/browser/

because android files will be a different and will need to add more files for Android.

There was a CL to move the browser files form bot/browser/ to /bot/ as i thought it will be the same files used for Android.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7119 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoRelanding https://code.google.com/p/webrtc/source/detail?r=7093, after it got
mallinath@webrtc.org [Tue, 9 Sep 2014 14:38:10 +0000 (14:38 +0000)]
Relanding https://code.google.com/p/webrtc/source/detail?r=7093, after it got
reverted due to some internal compile failures.

In this CL changes are done in portallocator_unittest.cc, in particular to EXPECT_EQ checking in new tests.

Original patch committed in https://code.google.com/p/webrtc/source/detail?r=7093

TBR=juberti@webrtc.org
BUG=1179

Review URL: https://webrtc-codereview.appspot.com/22329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7118 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agofix a bug in the logic when new Networks are merged. This happens when
guoweis@webrtc.org [Tue, 9 Sep 2014 13:54:45 +0000 (13:54 +0000)]
fix a bug in the logic when new Networks are merged. This happens when
we have 2 networks with the same key

BUG=410554 in chromium

http://code.google.com/p/chromium/issues/detail?id=410554

Corresponding change in chromium is
https://codereview.chromium.org/536133003/

R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19249005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7117 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoMore suppressions, uninitialized read in cricket::VideoFrame::Validate
sprang@webrtc.org [Tue, 9 Sep 2014 11:50:19 +0000 (11:50 +0000)]
More suppressions, uninitialized read in cricket::VideoFrame::Validate

BUG=3789
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7116 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoPeerconnection_jni to use webrtc/base/checks.h instead of implementing its own.
andresp@webrtc.org [Tue, 9 Sep 2014 11:45:44 +0000 (11:45 +0000)]
Peerconnection_jni to use webrtc/base/checks.h instead of implementing its own.

This needs to happen sooner or later as if webrtc/base/checks.h happens to be included transitively here it would collide.

R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7115 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoExpose VideoEncoders with webrtc/video_encoder.h.
pbos@webrtc.org [Tue, 9 Sep 2014 10:40:56 +0000 (10:40 +0000)]
Expose VideoEncoders with webrtc/video_encoder.h.

Exposes VideoEncoders as part of the public API and provides a factory
method for creating them.

BUG=3070
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7114 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoInitialize ChannelBuffer's memory to avoid uninitialized reads.
andrew@webrtc.org [Mon, 8 Sep 2014 23:11:44 +0000 (23:11 +0000)]
Initialize ChannelBuffer's memory to avoid uninitialized reads.

Removed the zero out memset in this change:
https://review.webrtc.org/24469004/

assuming it was unneeded. Dr. Memory taught me that assupmtion was
invalid. linux_memcheck try runs might have caught this, if they
weren't flaking out on unrelated stuff.

TBR=claguna@google.com

Review URL: https://webrtc-codereview.appspot.com/28429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7113 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoRevert 7093: "Implementing ICE Transports type handling in libjingle transport."
henrike@webrtc.org [Mon, 8 Sep 2014 22:46:28 +0000 (22:46 +0000)]
Revert 7093: "Implementing ICE Transports type handling in libjingle transport."

TBR=mallinath@webrtc.org
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/28419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7112 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoConvert GN visibility to be a list.
brettw@chromium.org [Mon, 8 Sep 2014 22:45:18 +0000 (22:45 +0000)]
Convert GN visibility to be a list.

GN visibility currently allows either string or list types, but this is causing
some problems for some templates. I'm going to require it to be lists, so am
changing all callers before pushing the new binary.

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7111 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoFinish work queue in SctpDataMediaChannelTest.
pbos@webrtc.org [Mon, 8 Sep 2014 21:44:07 +0000 (21:44 +0000)]
Finish work queue in SctpDataMediaChannelTest.

Always finishing the work queue prevents memory leak detected in
LeakSanitizer (packet is deleted on the receiver side).

R=jiayl@webrtc.org
BUG=3608,chromium:375154

Review URL: https://webrtc-codereview.appspot.com/28399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7110 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoFix a bot-breaking memory leak from early returning in ParseMediaDescription.
jiayl@webrtc.org [Mon, 8 Sep 2014 21:43:43 +0000 (21:43 +0000)]
Fix a bot-breaking memory leak from early returning in ParseMediaDescription.

BUG=3791
R=henrike@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7109 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoRevert "Fixes two issues in how we handle OfferToReceiveX for CreateOffer:" because...
jiayl@webrtc.org [Mon, 8 Sep 2014 20:44:36 +0000 (20:44 +0000)]
Revert "Fixes two issues in how we handle OfferToReceiveX for CreateOffer:" because it broke content_browsertests on Android.

This reverts commit r7068.

TBR=kjellander@webrtc.org
BUG=2108

Review URL: https://webrtc-codereview.appspot.com/23539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7108 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoAdd ctors to ChannelBuffer to enable copying on construction.
andrew@webrtc.org [Mon, 8 Sep 2014 20:27:04 +0000 (20:27 +0000)]
Add ctors to ChannelBuffer to enable copying on construction.

Also:
- Fix the constness of some parameters.
- Add more const overloads.
- Use DCHECK in place of assert.
- Removed an unnecessary memset.

R=claguna@google.com

Review URL: https://webrtc-codereview.appspot.com/24469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7107 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days ago(Auto)update libjingle 74955991-> 75042522
buildbot@webrtc.org [Mon, 8 Sep 2014 19:45:36 +0000 (19:45 +0000)]
(Auto)update libjingle 74955991-> 75042522

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7106 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoSuppress uninitialized read warning in cricket::VideoFrame::Validate
sprang@webrtc.org [Mon, 8 Sep 2014 14:00:38 +0000 (14:00 +0000)]
Suppress uninitialized read warning in cricket::VideoFrame::Validate

BUG=3789
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7105 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoSet a default speech type in iSAC wrapper
henrik.lundin@webrtc.org [Mon, 8 Sep 2014 13:40:58 +0000 (13:40 +0000)]
Set a default speech type in iSAC wrapper

If the decoder encounters an error, it may leave the speech type
unassigned, leading to a use-of-uninitialized-value in subsequent lines.

BUG=crbug/411162
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7104 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoStarting to implement the new ACM API
henrik.lundin@webrtc.org [Mon, 8 Sep 2014 13:13:19 +0000 (13:13 +0000)]
Starting to implement the new ACM API

The new implementation class is called AudioCodingImpl, and will in the
end replace AudioCodingModuleImpl.

This is work in progress.

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7103 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoAdding the ability to test on Chrome for Android.
houssainy@google.com [Mon, 8 Sep 2014 13:01:40 +0000 (13:01 +0000)]
Adding the ability to test on Chrome for Android.
use "android-chrome" as type in rtcbot running command.
Example: node test.js android-chrome

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7102 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoaudio_processing: Removed use of macro WEBRTC_SPL_UMUL_16_16
bjornv@webrtc.org [Mon, 8 Sep 2014 11:21:56 +0000 (11:21 +0000)]
audio_processing: Removed use of macro WEBRTC_SPL_UMUL_16_16

The macro replaced is a trivial multiplication after explicit casts to uint16_t and uint32_t. This CL replaces its use with "*" and adds explicit casts if necessary.

Affected components:
* AECMobile
* AGC
* Noise Suppression (fixed point version)

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7101 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agovideo_processing: Removed usage of WEBRTC_SPL_UMUL_16_16
bjornv@webrtc.org [Mon, 8 Sep 2014 11:19:39 +0000 (11:19 +0000)]
video_processing: Removed usage of WEBRTC_SPL_UMUL_16_16

The trivial macro WEBRTC_SPL_UMUL_16_16 is nothing but plain mutliplication of casted values. This CL explicitly use "*" at place and casts if necessary.

BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7100 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days ago- Adding AndroidDeviceManager to botManager.js to help in selecting devices, in case...
houssainy@google.com [Mon, 8 Sep 2014 10:36:11 +0000 (10:36 +0000)]
- Adding AndroidDeviceManager to botManager.js to help in selecting devices, in case running test on Android devices.

- Select BotType using nodeJs terminal command.

- ping_pong.js test added.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7099 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoRoll chromium_revision 94532b1..ea769fd
kjellander@webrtc.org [Mon, 8 Sep 2014 10:06:37 +0000 (10:06 +0000)]
Roll chromium_revision 94532b1..ea769fd

Summary of changes (git diff 94532b1..ea769fd DEPS):
* buildtools 2328da4..ea4dc0e
* third_party/android_tools 3186999..7fc902d
* third_party/boringssl 6c7aed0..7bdec13
* third_party/libjpeg_turbo 2ed5319..3963fbc
* third_party/libvpx 982d147..ceebbcc (r291730:291805)
* third_party/nss 90c5f9a..7b5b6ec
* third_party/usrsctp/usrsctplib e6e1833..8975bd5

BUG=3608
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7098 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoFix RTT calculations for send-only channels.
stefan@webrtc.org [Mon, 8 Sep 2014 08:45:25 +0000 (08:45 +0000)]
Fix RTT calculations for send-only channels.

As we don't know the SSRC of the other end in a send-only channel since we haven't received packets from that end, we are required to assume that the SSRC of the first report block is the correct SSRC to use for RTT calculations.

BUG=3781
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7097 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoIgnore FEC packet in stats, if it is first packet on ssrc.
sprang@webrtc.org [Mon, 8 Sep 2014 08:20:18 +0000 (08:20 +0000)]
Ignore FEC packet in stats, if it is first packet on ssrc.

BUG=chrome:410456
R=mflodman@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7096 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoGN: Prefix WebRTC specific variables with "rtc_"
kjellander@webrtc.org [Sun, 7 Sep 2014 17:36:10 +0000 (17:36 +0000)]
GN: Prefix WebRTC specific variables with "rtc_"

BUG=3441
TESTED=Trybots + Running GN in a Chromium checkout with
src/third_party/webrtc symlinked to the WebRTC checkout
with this CL applied, both with the default GN settings
and using: --args="os=\"android\" cpu_arch=\"arm\""

R=brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/27379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7095 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoAdd video_capture_tests_apk_target
kjellander@webrtc.org [Sun, 7 Sep 2014 17:35:51 +0000 (17:35 +0000)]
Add video_capture_tests_apk_target

In https://codereview.chromium.org/500423004/ the
target that was previously used to build the Android APK
tests was removed. When building these tests from a
standalone checkout, the video_capture_tests_apk target
was missing in the chain of targets that gets generated
into the 'all' target.

BUG=3764
TESTED=Trybots.
TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7094 4adac7df-926f-26a2-2b94-8c16560cd09d