Commit Graph

812 Commits

Author SHA1 Message Date
guoweis@webrtc.org
97ed39344a Reapply 23529005 after fixing the build break issue (Chromium:582133002)
Review URL: https://webrtc-codereview.appspot.com/23529005

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

-

R=tkchin@webrtc.org

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

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

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

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

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

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7244 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-19 16:53:46 +00:00
pbos@webrtc.org
bbe0a8517d Config struct for VideoEncoder.
Used for config parameters in common between multiple codecs as well as
the encoder-specific pointer. In particular this contains content mode
(realtime video vs. screenshare).

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

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7239 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-19 12:30:25 +00:00
buildbot@webrtc.org
6e5c78422d (Auto)update libjingle 75875619-> 75878731
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7235 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-19 06:46:37 +00:00
buildbot@webrtc.org
b5a5c44ef7 (Auto)update libjingle 75865376-> 75875619
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7234 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-19 05:36:18 +00:00
buildbot@webrtc.org
d7acf11e8d (Auto)update libjingle 75854833-> 75865376
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7233 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-19 02:01:09 +00:00
buildbot@webrtc.org
ccb3e3f3db (Auto)update libjingle 75854418-> 75854833
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7232 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-18 23:31:03 +00:00
buildbot@webrtc.org
dcc1f0426b (Auto)update libjingle 75852725-> 75853560
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7231 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-18 23:14:12 +00:00
glaznev@webrtc.org
0b435ba597 A few fixes to avoid crash in HW codec on device orientation change.
- Fix video encoder Reset() function to avoid setting codec
resolution to zero.
- Follow SW codec implementation and do not crash when frame
with the resolution different from the encoder resolution arrives.
Instead wait for at least 3 frames with new resolution and
re-initialize the codec. HW codec reset may take much longer
than SW codec, so these 3 frames threshold avoids resetting
codec when outstanding camera frame captured from previous device
orientation arrives.
- Plus some minor changes to make encoder reset/release
implementation closer to decoder implementation.

BUG=
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7230 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-18 23:01:03 +00:00
glaznev@webrtc.org
83af77bf3c Revert maximum video codec resolution on Android back to 720p again.
Some low end Android devices still have problems with 1080p support.

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

Review URL: https://webrtc-codereview.appspot.com/24639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7228 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-18 21:11:29 +00:00
buildbot@webrtc.org
933d88af58 (Auto)update libjingle 75818332-> 75837294
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7227 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-18 20:23:05 +00:00
jiayl@webrtc.org
42731bdded Avoid writing a double/float to a string to avoid a crash.
BUG=crbug/367223
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7225 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-18 16:51:51 +00:00
pbos@webrtc.org
6cd6ba8ae0 Expose VP8/H264 defaults through video_encoder.h.
Reduces code duplication quite a bit, these identical defaults were set
in quite a few different places.

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

Review URL: https://webrtc-codereview.appspot.com/19299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7220 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-18 12:42:28 +00:00
andresp@webrtc.org
ab071daab8 Split video_render_module implementation into default and internal implementation.
Targets must now link with implementation of their choice instead of at "gyp"-time.

Targets linking with libjingle_media:
- internal implementation when build_with_chromium=0, default otherwise.

Targets linking with default render implementation:
- video_engine_tests
- video_loopback
- video_replay
- anything dependent on webrtc_test_common

Targets linking with internal render implementation:
- vie_auto_test
- video_render_tests
- libwebrtcdemo-jni
- video_engine_core_unittests

GN changes:
- Not many since there is almost no test definitions.

Work-around for chromium:
- Until chromium has updated libpeerconnection to link with video_capture_impl and video_render_impl, webrtc target automatically depends on it. This should fix the FYI bots and not require a webrtc roll to fix.

Re-enable android tests by reverting 7026 (some tests left disabled).

TESTED: passes all the bots. If this inadvertently breaks a target please fix the linking rules so the target has the desired implementation linked in.
BUG=3770
R=kjellander@webrtc.org, pbos@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7217 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-18 08:58:15 +00:00
guoweis@webrtc.org
369a637ac8 Implemented Network::GetBestIP() selection logic as following.
1) return the first global temporary and non-deprecrated ones.
2) if  not available, return global one.
3) if  not available, use ULA ipv6 as last resort.

ULA stands for unique local address. They are only useful in a private
WebRTC deployment. More detail: http://en.wikipedia.org/wiki/Unique_local_address

BUG=3808

At this point, rule  actually won't happen at current
implementation. The reason being that ULA address starting with 0xfc 0r 0xfd will be grouped into its own Network. The result of that is WebRTC will have one extra Network to generate candidates but the lack of rule  shouldn't prevent turning on IPv6 since ULA should only be tried in a close deployment anyway.

R=jiayl@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7200

Committed: https://code.google.com/p/webrtc/source/detail?r=7201

Review URL: https://webrtc-codereview.appspot.com/31369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7216 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17 22:37:29 +00:00
glaznev@webrtc.org
3b67f8e0ca Enable HW video decoding on Qualcomm devices.
Parallel decoding and encoding problem is fixed now
(b/16353967), so it is possible to start using Qualcomm
VP8 HW decoder. Bitrate overshoots should be fixed as well.

BUG=
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7215 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17 21:25:51 +00:00
henrike@webrtc.org
4a5061fbff talk/p2p/base: removed unused variable "port_"
BUG=N/A
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7212 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17 12:33:07 +00:00
andresp@webrtc.org
a74eda1b6f Split video_capture_module specific implementation (external vs internal capture)
into its own targets. Dependencies must link directly with the desired one.

Targets linking with libjingle_media:
 - internal implementation when build_with_chromium=0, default otherwise.

Targets linking with default/external capture implementation:
 - anything dependent on webrtc_test_common
 - anything dependent on video_engine_core

Targets linking with internal capture implementation:
 - vie_auto_test
 - anything dependent on webrtc_test_renderer

GN changes:
 - Not many since there is almost no test definitions.

TESTED: passes all the bots. If this inadvertently breaks a target please fix the linking rules so the target has the desired implementation linked in.

BUG=3768
R=glaznev@webrtc.org
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7209 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17 11:50:19 +00:00
andresp@webrtc.org
85ef770d92 Split video engine android initialization into each internal module initialization.
This is to later on allow targets to pick at link time if to include the external or internal implementation. In order to do that the video_engine cannot compile different based on which option is picked later on.

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

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7208 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17 11:44:51 +00:00
pbos@webrtc.org
ab990ae43a Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h.""
Re-lands r7114 after landing r7204 to adress the compile error causing
the rollback in r7151.

BUG=3070
TBR=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7207 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17 09:02:25 +00:00
buildbot@webrtc.org
6a9b155798 (Auto)update libjingle 75683337-> 75695882
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7206 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17 08:08:38 +00:00
glaznev@webrtc.org
a59c501c99 Java VideoRenderer class may be backed by two different native
classes depending on type of rendering.
Fix crash in AppRtcDemo by calling correct destructor on exit.

BUG=
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7202 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17 03:26:59 +00:00
guoweis@webrtc.org
40c2aa36f2 Implemented Network::GetBestIP() selection logic as following.
1) return the first global temporary and non-deprecrated ones.
2) if  not available, return global one.
3) if  not available, use ULA ipv6 as last resort.

ULA stands for unique local address. They are only useful in a private
WebRTC deployment. More detail: http://en.wikipedia.org/wiki/Unique_local_address

BUG=3808

At this point, rule  actually won't happen at current
implementation. The reason being that ULA address starting with 0xfc 0r 0xfd will be grouped into its own Network. The result of that is WebRTC will have one extra Network to generate candidates but the lack of rule  shouldn't prevent turning on IPv6 since ULA should only be tried in a close deployment anyway.

R=jiayl@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7200

Review URL: https://webrtc-codereview.appspot.com/31369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7201 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-16 20:29:41 +00:00
guoweis@webrtc.org
f8bff762d1 Implemented Network::GetBestIP() selection logic as following.
1) return the first global temporary and non-deprecrated ones.
2) if  not available, return global one.
3) if  not available, use ULA ipv6 as last resort.

ULA stands for unique local address. They are only useful in a private
WebRTC deployment. More detail: http://en.wikipedia.org/wiki/Unique_local_address

BUG=3808

At this point, rule  actually won't happen at current
implementation. The reason being that ULA address starting with 0xfc 0r 0xfd will be grouped into its own Network. The result of that is WebRTC will have one extra Network to generate candidates but the lack of rule  shouldn't prevent turning on IPv6 since ULA should only be tried in a close deployment anyway.

R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7200 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-16 20:17:22 +00:00
pbos@webrtc.org
cddd17c0f8 Recreate VideoStreams when setting resolution.
Instead of just changing resolution on the last stream streams are
reallocated to make sure that all streams are updated to match the
new input resolution.

R=pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/29469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7197 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-16 16:33:13 +00:00
pbos@webrtc.org
88e85ad64d Add pbos@webrtc.org (myself) to talk/media/webrtc/.
Allows easier reviews of webrtcvideoengine2.cc and landing the new video
API on shorter review cycles.

R=pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/30409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7196 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-16 16:14:51 +00:00
buildbot@webrtc.org
80132e4d70 (Auto)update libjingle 75610402-> 75610402
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7194 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-16 15:24:15 +00:00
kjellander@webrtc.org
595b23c66f 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
2014-09-16 08:58:22 +00:00
andrew@webrtc.org
6ae5a6d7fe 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
2014-09-16 01:03:29 +00:00
glaznev@webrtc.org
996784548d 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
2014-09-15 17:52:42 +00:00
guoweis@webrtc.org
cd309e3168 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
2014-09-15 16:31:13 +00:00
pbos@webrtc.org
000d86792d 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
2014-09-15 14:38:07 +00:00
henrike@webrtc.org
7f826350e3 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
2014-09-15 08:13:36 +00:00
buildbot@webrtc.org
a42a3ade54 (Auto)update libjingle 75390072-> 75428737
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7174 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-13 01:09:18 +00:00
fbarchard@google.com
7e31197cb2 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
2014-09-13 00:52:42 +00:00
glaznev@webrtc.org
192a54ff2f 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
2014-09-12 16:40:35 +00:00
henrike@webrtc.org
3decd9b776 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
2014-09-12 16:31:29 +00:00
buildbot@webrtc.org
ea77334c30 (Auto)update libjingle 75302540-> 75327856
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7160 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 21:52:48 +00:00
henrike@webrtc.org
1d8f780779 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
2014-09-11 17:16:56 +00:00
glaznev@webrtc.org
1d53f64b0f 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
2014-09-11 16:58:25 +00:00
henrikg@webrtc.org
307d3dbdee 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
2014-09-11 09:48:30 +00:00
sprang@webrtc.org
c665dcb205 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
2014-09-11 08:29:53 +00:00
henrik.lundin@webrtc.org
1972ff8a6e 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
2014-09-11 06:20:28 +00:00
henrike@webrtc.org
4c876453c8 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
2014-09-10 22:18:04 +00:00
glaznev@webrtc.org
3472dcd7b0 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
2014-09-10 19:24:57 +00:00
henrike@webrtc.org
b2efb6771c 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
2014-09-10 17:28:19 +00:00
jiayl@webrtc.org
b6d69282f5 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
2014-09-10 16:31:34 +00:00
buildbot@webrtc.org
5d639b3ef3 (Auto)update libjingle 75141932-> 75179475
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7129 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10 07:57:12 +00:00
jiayl@webrtc.org
7d4891d3f1 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
2014-09-09 21:43:15 +00:00
fbarchard@google.com
54cf1505e2 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
2014-09-09 18:34:53 +00:00
jiayl@webrtc.org
22406fcc9b 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
2014-09-09 15:44:05 +00:00
mallinath@webrtc.org
3d81b1b22a 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
2014-09-09 14:38:10 +00:00
andresp@webrtc.org
4d19e05ab2 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
2014-09-09 11:45:44 +00:00
pbos@webrtc.org
b420191743 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
2014-09-09 10:40:56 +00:00
henrike@webrtc.org
8b0b21161a 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
2014-09-08 22:46:28 +00:00
pbos@webrtc.org
7118e61669 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
2014-09-08 21:44:07 +00:00
jiayl@webrtc.org
0e52772aa9 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
2014-09-08 21:43:43 +00:00
jiayl@webrtc.org
c172320bd2 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
2014-09-08 20:44:36 +00:00
buildbot@webrtc.org
fd42f9dd6f (Auto)update libjingle 74955991-> 75042522
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7106 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 19:45:36 +00:00
mallinath@webrtc.org
7256d31d28 Implementing ICE Transports type handling in libjingle transport.
BUG=1179
R=juberti@webrtc.org, bemasc@webrtc.org, jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7093 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-07 04:08:44 +00:00
thorcarpenter@google.com
cc060563f3 Remove unnecessary include from testutils.cc.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7090 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 21:19:00 +00:00
buildbot@webrtc.org
992febb997 (Auto)update libjingle 74873066-> 74873164
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7089 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 16:39:08 +00:00
thorcarpenter@google.com
a3344cfda4 Fix webrtcvideoframe tests.
R=fbarchard@google.com, harryjin@google.com, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7088 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 16:34:13 +00:00
jiayl@webrtc.org
ddb85ab85b Updated SCTP SDP attributes according to draft-ietf-mmusic-sctp-sdp-07
- SDP sctpmap attribute replaced with fmtp attribute
- SDP sctp-port attribute is newly added

BUG=3592
R=jiayl@webrtc.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7087 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 16:31:56 +00:00
buildbot@webrtc.org
af5fa95258 (Auto)update libjingle 74857067-> 74860820
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7084 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 13:03:50 +00:00
buildbot@webrtc.org
7e3bd3d7de (Auto)update libjingle 74851128-> 74857067
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7083 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 11:45:42 +00:00
buildbot@webrtc.org
bc6fa1876e (Auto)update libjingle 74825992-> 74851128
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7082 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 11:08:01 +00:00
buildbot@webrtc.org
818b7b3ac9 (Auto)update libjingle 74825084-> 74825992
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7074 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 00:14:03 +00:00
jiayl@webrtc.org
dfbcf8161e Fix an issue in MediaStreamSignaling that a remotely create DataChannel is added to the list twice.
BUG=3778
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7073 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 00:01:12 +00:00
henrike@webrtc.org
f1427c6731 Revert 7070 "TurnPort should retry allocation with a new address on error
STUN_ERROR_ALLOCATION_MISMATCH."

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

Review URL: https://webrtc-codereview.appspot.com/15359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7072 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 22:21:33 +00:00
glaznev@webrtc.org
4b234044d5 Reduce maximum video resolution for Android.
HW video encoder and decoder can not be initialized
with 3840x2160 resolution.

BUG=3757,3738
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7071 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 19:50:07 +00:00
jiayl@webrtc.org
574f2f60fe TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH.
BUG=3570
R=juberti@webrtc.org, mallinath@webrtc.org

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7070 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 19:11:34 +00:00
jiayl@webrtc.org
52055a276d 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

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7068 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 17:12:25 +00:00
pbos@webrtc.org
ceb956b29d Abort Negotiate() if DoCreateOffer() fails.
Addressing crash in test.

R=jiayl@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/19239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7066 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 15:27:49 +00:00
pbos@webrtc.org
bcb6bcfe6c Remove HybridVideoEngine.
This is currently unused dead code.

R=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/24409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7055 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 07:32:26 +00:00
thorcarpenter@google.com
95c2458766 * Move test data assests required by video frame tests to be in libjingle instead of elsewhere and co-located with other libjingle test data files.
"gcl try" fails to upload these large files so adding them independently.

R=andrew@webrtc.org, harryjin@google.com, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7050 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 23:17:36 +00:00
buildbot@webrtc.org
609f987488 (Auto)update libjingle 74696326-> 74723281
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7047 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 21:50:32 +00:00
buildbot@webrtc.org
fa4535b270 (Auto)update libjingle 74694022-> 74696326
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7045 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 16:49:04 +00:00
pbos@webrtc.org
26c0c41a06 Network up/down signaling in Call.
BUG=2429
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13109005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7044 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 16:17:12 +00:00
pbos@webrtc.org
ebee401230 Remove flake in SendsLowerResolutionOnSmallerFrames.
Speculative fix for break on Linux64 Release. It looks like the second
frame is being dropped which is likely because the two frames are sent
too close to eachother. Adding a delay of 33ms in between them to make
sure the second one isn't dropped.

R=minyue@webrtc.org
TBR=pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/22289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7043 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 15:52:02 +00:00
pbos@webrtc.org
c4175b9fdf Set resolution based on incoming VideoFrames.
R=pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/17269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7042 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 15:25:49 +00:00
buildbot@webrtc.org
72e448559d (Auto)update libjingle 74628537-> 74648573
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7033 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 00:43:48 +00:00
tkchin@webrtc.org
90750482fa Remove deprecated RTCVideoRenderer constructor.
Removes -[RTCVideoRenderer initWithView]. Also, fix potential issue where we hold on to a video frame longer than the lifetime of its associated track.

BUG=3341
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7032 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02 20:50:00 +00:00
pbos@webrtc.org
9f341283f6 Remove WebRtcVideoEngine::default_codec_format().
R=pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/24399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7029 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02 16:33:09 +00:00
pbos@webrtc.org
03655143db Remove files from talk/PRESUBMIT.py.
BUG=
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23429005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7028 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02 16:17:36 +00:00
thakis@chromium.org
44010f3e52 win: Replace custom assert() macro with regular assert.h
The current code got added in libjingle r103; I don't see a good reason for it.
Things still build with plain old assert.h.

The custom assert was wrong: __debugbreak() is documented to return void,
so doing `cond ? true : __debugbreak()` shouldn't build (and it doesn't in
clang-cl). It's possible to make it build by writing
`cond ? true : (__debugbreak(), true)`, but just using the regular header
seems like a much better fix.

BUG=chromium:82385
Review URL: https://webrtc-codereview.appspot.com/19139004/


git-svn-id: http://webrtc.googlecode.com/svn/trunk@7007 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-29 03:00:15 +00:00
jiayl@webrtc.org
bc3f333905 Add jiayl to talk OWNERS.
BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7006 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-28 23:24:36 +00:00
jiayl@webrtc.org
e21cc9ae2a When the peerconnection creates the offer with a constraint to disable the audio offering, stats will not get properly updated.
constraints . SetMandatoryReceiveAudio (false);

The problem is that webrtc::GetTrackIdBySsrc returns false if audio is not available. However it should continue and check for the video track.

BUG=webrtc:3755
R=jiayl@webrtc.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7005 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-28 22:21:34 +00:00
niklas.enbom@webrtc.org
4431fd6ad5 Add 60 fps video support
R=henrike@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7000 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-28 14:57:46 +00:00
buildbot@webrtc.org
1f8a23757a (Auto)update libjingle 74235596-> 74297316
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6997 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-28 10:52:44 +00:00
pbos@webrtc.org
75c3ec1763 Fix data races during VideoAdapterTest tear-down.
Explicitly disconnect the VideoCapturer to avoid frames being
delivered during listener destruction. This manifested only on DrMemory
Full on Windows which I was able to repro locally.

BUG=3671
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6991 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-27 18:16:13 +00:00
buildbot@webrtc.org
573a1eef3d (Auto)update libjingle 74202294-> 74230205
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6990 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-27 17:21:19 +00:00
solenberg@webrtc.org
00f11f5e24 - Make local constant non-static.
- Remove spammy log line.

BUG=
R=henrike@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6987 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-27 08:52:17 +00:00
guoweis@webrtc.org
7087857afd implement handling ALTERNATE-SERVER response from turn protocol as
specified in RFC 5766, also created 2 test cases for both the normal
redirection case as well as when a pingpong situation happens, the
allocation should fail

BUG=1986 TURN ALTERNATE-SERVER support
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6985 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-26 21:37:49 +00:00
buildbot@webrtc.org
3533bfcb94 (Auto)update libjingle 74132319-> 74133664
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6983 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-26 15:50:23 +00:00