0d523eea83
Remove static initializer from WebRtcVideoEngine2.
...
BUG=
R=pliard@google.com , pthatcher@webrtc.org , pliard@chromium.org
Review URL: https://webrtc-codereview.appspot.com/15679005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6338 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-05 09:10:55 +00:00
f1adbeedb4
(Auto)update libjingle 68562943-> 68571194
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6333 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-04 21:57:16 +00:00
738df8913d
Fix retain cycle in RTCEAGLVideoView.
...
CADisplayLink increases its target's refcount. In order to break retain cycle, we wrap CADisplayLink in a new RTCDisplayLinkTimer class and use that instead.
R=fischman@webrtc.org , noahric@chromium.org
BUG=3391
Review URL: https://webrtc-codereview.appspot.com/16599006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6331 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-04 20:19:39 +00:00
6f237769b3
(Auto)update libjingle 68507189-> 68543735
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6329 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-04 16:23:10 +00:00
40b45fc07a
(Auto)update libjingle 68506654-> 68507189
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6328 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-04 14:48:33 +00:00
0cdcd23a03
(Auto)update libjingle 68501302-> 68506654
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6321 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-04 01:31:14 +00:00
af81b9bffd
(Auto)update libjingle 68499439-> 68501302
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6320 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-04 00:08:54 +00:00
251fdf64cb
(Auto)update libjingle 68495561-> 68499439
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6319 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-03 23:43:48 +00:00
09a71cd9ce
talk/ios: Fixes source after corrupt sync in r6305 (which corrupted r6291).
...
BUG=N/A
R=tkchin@webrtc.org
TBR=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6318 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-03 22:46:23 +00:00
53217848b2
(Auto)update libjingle 68465410-> 68487517
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6317 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-03 21:09:11 +00:00
83eb7dff5c
PeerConnection(java): disable wait for flaky ICEConnection.COMPLETED.
...
This should be reverted when COMPLETED is delivered reliably.
BUG=3021
TESTED=without this patch the test fails in Debug mode after a handful of runs. With this patch 100 runs passed in a row on my desktop.
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6315 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-03 16:38:08 +00:00
289a35c56d
Add empty webrtcmediaengine.cc.
...
Should contain CreateWebRtcMediaEngine as soon as
libjingle/libjingle.gyp in Chromium builds this file. This file is added
ahead of time to get a smoother rolling process.
BUG=1788
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19599005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6313 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-03 14:51:34 +00:00
b525a9d790
(Auto)update libjingle 68379861-> 68445177
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6309 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-03 09:42:15 +00:00
044bdacfef
Remove kMaxWaitForStatsMs from tsanv2 compilation.
...
As some tests are #ifdef'd out on THREAD_SANITIZER this constant
triggers an unused-const-variable warning which breaks the build.
BUG=1205,3220
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6308 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-03 09:40:01 +00:00
34a08b4fb8
(Auto)update libjingle 68275107-> 68379861
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6305 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-02 15:48:10 +00:00
174a67439b
Enable -Wall, -Wextra and -Wunused-variable for talk/ on clang.
...
Also removes one case of unused-variable.
BUG=3220
R=henrike@webrtc.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15619005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6297 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-02 07:58:30 +00:00
8a09af3f67
Fix the build error from OpenSSLStreamAdapter::SSLVerifyCallback
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TBR=pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/17639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6296 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-30 23:24:08 +00:00
0163674f99
Make OpenSSLStreamAdapter verify the leaf certificate digest for chained certificates.
...
It used to compre a parent certificate's digest against the SDP fingerprint and caused connection failure.
BUG=3383
R=bemasc@webrtc.org , juberti@webrtc.org , rsleevi@chromium.org
Review URL: https://webrtc-codereview.appspot.com/17589005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6294 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-30 23:14:08 +00:00
56d114627b
Fix AppRTC target configuration in libjingle_examples.gyp.
...
libjingle_peerconnection_objc doesn't exist as a target in 32bit, so AppRTCDemo
needs that guard as well.
R=andrew@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/18489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6292 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-30 23:04:39 +00:00
acca675bcf
Implement mac version of AppRTCDemo.
...
- Refactored and moved AppRTCDemo to support sharing AppRTC connection code between iOS and mac counterparts.
- Refactored OpenGL rendering code to be shared between iOS and mac counterparts.
- iOS AppRTCDemo now respects video aspect ratio.
BUG=2168
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6291 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-30 22:26:06 +00:00
9f8164c060
Fix two bugs in DataChannel state transition.
...
1. OnStateChange should not be fired if state is not changed.
2. RemotePeerRequestClose should be a no-op if it's already closed.
TBR=pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/21559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6290 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-30 21:53:17 +00:00
1678db9df6
(Auto)update libjingle 68230113-> 68244456
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6287 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-30 14:02:09 +00:00
540a2251aa
(Auto)update libjingle 68230011-> 68230113
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6281 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-30 07:40:35 +00:00
35efb839ed
Implement new-API test RecvStreamWithoutRtx.
...
R=pthatcher@google.com , pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/20449005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6280 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-30 07:40:04 +00:00
c34bb3a886
Log default receive stream creation.
...
Log when receiving a packet that doesn't have a receiver, this way you
can tell from logs where the AddRecvStream call came from.
R=pthatcher@google.com , pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/17459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6279 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-30 07:38:43 +00:00
198647473b
Implement and fix new-API NackIsEnabled test.
...
Required enabling NACK on receiver side which was apparently missed.
BUG=1788
R=pthatcher@google.com , pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16499007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6278 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-30 07:35:47 +00:00
1d66be22c8
(Auto)update libjingle 68203780-> 68206793
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6277 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-29 22:54:24 +00:00
8dcd43c4f7
Make MediaSessionDescriptionFactory accept offers with UDP/TLS/RTP/SAVPF.
...
This is the first step toward switching completely to UDP/TLS/RTP/SAVPF.
BUG=2796
R=juberti@webrtc.org , pthatcher@google.com
Review URL: https://webrtc-codereview.appspot.com/13439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6276 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-29 22:07:59 +00:00
abe01dd634
AppRTCDemo(android): run in full-screen & immersive mode.
...
Also:
- Only show stats HUD on demand
- Only collect stats when HUD is showing
- Don't render solid green frame when video is not present in either direction
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6275 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-29 21:46:52 +00:00
5dc51fbe50
Closes the DataChannel when the send buffer is full or on transport errors.
...
As stated in the spec.
BUG=2645
R=pthatcher@google.com , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6270 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-29 15:33:54 +00:00
001fd2d503
Fire OnRenegotiationNeeded only for the first SCTP DataChannel.
...
Subsequent DataChannels do not need renegotiation since SCTP data streams are not negotiated through SDP.
BUG=2431
R=pthatcher@google.com , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6268 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-29 15:31:11 +00:00
43a1395370
AppRTCDemo(android): README updates for a shrinking envsetup.sh world.
...
There was duplicated (and out of date!) information in README relative to
getting-started so de-duped to point to getting-started as the canonical
reference.
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15589006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6265 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-28 17:29:09 +00:00
b364016cbb
Revert r6161 "Drop the DataChannel message if it's received when the channel is not open."
...
The spec does not say the DataChannel has to be open to receive a message.
TBR=pthatcher@google.com
BUG=crbug/363005
Review URL: https://webrtc-codereview.appspot.com/16569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6264 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-28 16:37:25 +00:00
f666ecc60d
Disabling flaky libjingle tests after fixit week.
...
BUG=webrtc:3316,webrtc:3317,webrtc:3318
TBR=fischman@google.com
Review URL: https://webrtc-codereview.appspot.com/12569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6250 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-27 08:08:00 +00:00
727ff69829
(Auto)update libjingle 67872893-> 67873348
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6244 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 23:20:53 +00:00
75cb3dc5f2
(Auto)update libjingle 67869540-> 67872893
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6243 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 23:13:35 +00:00
b445f26f24
Fixing correct UMA metric for PeerConnection enabled with IPv4 Vs IPv6.
...
BUG=N/A
TBR=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21499007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6242 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 22:19:37 +00:00
39eccefbde
Disable ChannelManagerTest.StartupShutdownOnUnstartedThread
...
The test is testing a scenario that shouldn't happen.
BUG=3388
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21509005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6238 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 17:50:38 +00:00
7aa1a4767f
(Auto)update libjingle 67848628-> 67848776
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6237 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 17:33:05 +00:00
e5063b1733
Thread: delete racy API (Release()) and fix racy code (started()).
...
- Thread::Release() wrote a local variable on the calling thread but read it on
another thread, with no synchronization. Happily it has no non-test callers
so deleting it instead of trying to fix it (see bug for details).
- Thread::started_ similarly was racily being written to; replaced with a
running_ Event, and hid the accessor except for tests & legacy callers,
with a note about why it's a bad idea.
webrtc/base patched with:
git diff origin --relative=talk/base | patch -p1 -dwebrtc/base
followed by manual merge of 3 thunks that ran afoul of naming differences
between talk/base and webrtc/base.
BUG=3388
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14589005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6236 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 17:28:50 +00:00
18f41b8eb4
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
...
Example files that this makes ok instead of flagging include:
talk/base/signalthread_unittest.cc
talk/base/thread_unittest.cc
webrtc/base/signalthread_unittest.cc
webrtc/base/thread.cc
webrtc/base/thread.h
webrtc/base/thread_unittest.cc
BUG=1027
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19539006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6235 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 17:27:18 +00:00
706152dcc9
Fix uninitialized reads in IsDefaultBrowserFirefox
...
BUG=
TEST=Local DrMemory.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19529006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6232 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 14:35:48 +00:00
8e755c1ad2
Connect SignalDestroyed in AllocationSequence after TURN ports are destroyed
...
when TURN ports are using shared socket with UDP port.
This is required as AllocationSequence maintains a map of turn ports. If the
ports are destroyed without the knowledge of AllocationSequence, sequence will
try to deliver packets to the destoyed ports.
R=jiayl@webrtc.org
BUG=https://code.google.com/p/chromium/issues/detail?id=368877
Review URL: https://webrtc-codereview.appspot.com/14569007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6219 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 23:00:46 +00:00
f9f1bfbdae
(Auto)update libjingle 67686255-> 67689476
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6216 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 17:02:15 +00:00
ce4201df52
(Auto)update libjingle 67643194-> 67686255
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6214 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 16:22:51 +00:00
000658a138
Revert of 6211 as it was committed despite of PRESUBMIT.py warning. The commit breaks the sync bot.
...
BUG=N/A
TBR=mcasas@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21519006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6212 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 16:01:13 +00:00
3b7e282caa
Disabling systematically failing
...
WebRtcVideoMediaChannelTest.SendVp8HdAndReceiveAdaptedVp8Vga
TBR= pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14569006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6211 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 14:25:20 +00:00
49a6a27bf0
(Auto)update libjingle 67555838-> 67643194
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6206 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 00:24:54 +00:00
1732a591e7
Add a UIView for rendering a video track.
...
RTCEAGLVideoView provides functionality to render a supplied RTCVideoTrack using OpenGLES2.
R=fischman@webrtc.org
BUG=3188
Review URL: https://webrtc-codereview.appspot.com/12489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6192 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-19 23:26:01 +00:00
40bc7779aa
talk_base: remove lock inversion between MessageQueue and MessageQueueManager.
...
Removes the concept of a MessageQueue being "active" in favor of considering all
live MQ's to be active.
(previously a MQ was active starting from the first Post to it and stopped being
active in its dtor).
BUG=3230
R=sriniv@google.com
Review URL: https://webrtc-codereview.appspot.com/21489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6190 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-19 17:58:04 +00:00
