buildbot@webrtc.org
437d57db5b
(Auto)update libjingle 71775619-> 71778545
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6771 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-23 21:40:28 +00:00
henrike@webrtc.org
8c7e3291a9
Revert 6747 "Refactor StatsCollector and associated types."
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Breakes FYI bots.
BUG=N/A
TBR=ajm@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6770 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-23 21:38:58 +00:00
henrike@webrtc.org
8721f989bf
Revert 6766 "Temporarily add a default ctor to StatsReport and make |id| non const. As soon as I've updated the chrome side, I'll revert this cl."
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BUG=N/A
TBR=ajm@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6769 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-23 21:38:09 +00:00
buildbot@webrtc.org
e2da234e27
(Auto)update libjingle 71766184-> 71775619
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6768 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-23 21:09:01 +00:00
tommi@webrtc.org
0f7328cd6b
Temporarily add a default ctor to StatsReport and make |id| non const.
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As soon as I've updated the chrome side, I'll revert this cl.
TBR=henrike
Review URL: https://webrtc-codereview.appspot.com/16149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6766 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-23 16:31:57 +00:00
glaznev@webrtc.org
efe4b9af49
Add VP8 video decoding hw acceleration support to Java Peerconnection library.
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For now NVidia decoder is supported only,
Qualcomm will be added once b/16353967 is fixed.
TODO:
- Support queuing 2-3 decoder input buffers.
- Add average decoding time statistics.
- Add Qualcomm hw decoder support.
BUG=3030
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6758 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-22 17:44:53 +00:00
buildbot@webrtc.org
e69b061926
(Auto)update libjingle 71575585-> 71599033
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6750 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-21 20:38:58 +00:00
tommi@webrtc.org
908f57ed94
Disable GetStatsForInvalidTrack while I rewrite it.
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TBR=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17969005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6748 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-21 11:44:39 +00:00
tommi@webrtc.org
756b8462eb
Refactor StatsCollector and associated types.
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* Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
* Reports are now managed in a set, not a map, since it's enough to store the id in one place.
* Report ids are now const.
* Copying of data has been greatly reduced.
* This change includes preparation work for making GetStats fully async.
R=xians@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=6745
Review URL: https://webrtc-codereview.appspot.com/18819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6747 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-21 11:24:17 +00:00
tommi@webrtc.org
fd61a1d693
Revert 6745 "Refactor StatsCollector and associated types."
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Broke build on android.
> Refactor StatsCollector and associated types.
> * Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
> * Reports are now managed in a set, not a map, since it's enough to store the id in one place.
> * Report ids are now const.
> * Copying of data has been greatly reduced.
> * This change includes preparation work for making GetStats fully async.
>
> R=xians@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/18819004
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6746 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-21 11:05:28 +00:00
tommi@webrtc.org
647e05cfcd
Refactor StatsCollector and associated types.
...
* Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
* Reports are now managed in a set, not a map, since it's enough to store the id in one place.
* Report ids are now const.
* Copying of data has been greatly reduced.
* This change includes preparation work for making GetStats fully async.
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6745 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-21 10:55:11 +00:00
jiayl@webrtc.org
3edbaaf337
Ignore empty data in DataChannel::Send to match FF's behavior.
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BUG=crbug/395205
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6742 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 23:57:50 +00:00
jiayl@webrtc.org
a0b929b63c
Revert "Reland r6707 with the fix for callclient.cc."
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Breaking pulse build again.
This reverts commit 3e0bb9b5bf7f616000399e24f1d9622ad6b612f9.
TBR=wu@webrtc.org
BUG=3310
Review URL: https://webrtc-codereview.appspot.com/17979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6740 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 22:28:36 +00:00
buildbot@webrtc.org
3dec81a736
(Auto)update libjingle 71456173-> 71456344
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6738 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 21:39:56 +00:00
jiayl@webrtc.org
a6e8cf8fb7
Reland r6707 with the fix for callclient.cc.
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TBR=mallinath@webrtc.org
BUG=3310
Review URL: https://webrtc-codereview.appspot.com/13039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6737 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 21:34:11 +00:00
buildbot@webrtc.org
60e65b11c1
(Auto)update libjingle 71452608-> 71453580
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6735 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 21:07:50 +00:00
jiayl@webrtc.org
8636fc852e
Creates the default track if the remote media content is send-only and there is no stream in the SDP.
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BUG=2628
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6734 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 20:54:27 +00:00
mallinath@webrtc.org
e5995aadd5
Assigning a priority to TURN server list passed to PeerConnection. First entry in the TURN server list will get the highest priotity and so forth.
...
This priority will be used in calculating the candidate priority generated from the server. This will allow candidate generated from server to have unique priority.
BUG=3223
R=jiayl@webrtc.org , juberti@webrtc.org , pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6721 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 18:23:52 +00:00
jiayl@webrtc.org
e10d28cf14
fix
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6720 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 17:07:49 +00:00
wu@webrtc.org
52eddec71b
Revert 6707 "Add support of multiple STUN servers in UDPPort."
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Reason:
Breaks the build on callclient.cc.
> Add support of multiple STUN servers in UDPPort.
> Now UDPPort signals PortComplete or PortError when the Bind requests for all STUN servers are responded or failed. If any STUN bind is successful, PortComplete is signaled; otherwise, PortError is signaled.
>
> I discovered a bug in SocketAddress while working on this. It didn't consider two addresses unequal if they have unresolved IP and different hosts. It's fixed now.
>
> BUG=3310
> R=mallinath@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/13879004
TBR=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6711 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 00:03:24 +00:00
wu@webrtc.org
4c3e9917e7
Make sure b lines appear before all the a lines. Per RFC 4566, the order of media description should be:
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m= (media name and transport address)
i=* (media title)
c=* (connection information -- optional if included at
session level)
b=* (zero or more bandwidth information lines)
k=* (encryption key)
a=* (zero or more media attribute lines)
BUG=2260
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6708 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-16 21:03:13 +00:00
jiayl@webrtc.org
46fb331bc5
Add support of multiple STUN servers in UDPPort.
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Now UDPPort signals PortComplete or PortError when the Bind requests for all STUN servers are responded or failed. If any STUN bind is successful, PortComplete is signaled; otherwise, PortError is signaled.
I discovered a bug in SocketAddress while working on this. It didn't consider two addresses unequal if they have unresolved IP and different hosts. It's fixed now.
BUG=3310
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6707 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-16 20:55:31 +00:00
tommi@webrtc.org
47218956fc
Minor refactoring of StatsCollector.
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* Make GetTimeNow a static method in the cc file.
* Make GetTransportIdFromProxy a static method as well and not a class method.
The second change is in preparation of removing the proxy_to_transport_ member variable which isn't needed and is just a copy from the session stats.
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6696 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-15 19:22:37 +00:00
tommi@webrtc.org
242068d58c
A step towards changing StatsReport::Value::name to an enum.
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The stats reporting code does a lot of unnecessary string copying.
This is a step in the direction of removing that and forcing use of only known constants.
This is a reland of an already reviewed cl that got reverted by mistake.
TBR=xians@google.com
Review URL: https://webrtc-codereview.appspot.com/12989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6678 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-14 20:19:56 +00:00
tommi@webrtc.org
03505bcb7a
Make StatsCollector depend on always having a valid session pointer.
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This is required since the session pointer is currently used on multiple threads but there's no synchronization code to guard it.
I'm removing the set_session() method and session() getter since they would cause problems if used without synchronization.
This is a reland of an already reviewed cl that got reverted by mistake.
TBR=xians@google.com
Review URL: https://webrtc-codereview.appspot.com/13959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6677 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-14 20:15:26 +00:00
buildbot@webrtc.org
d8524348bb
(Auto)update libjingle 71107853-> 71115715
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6675 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-14 20:05:09 +00:00
buildbot@webrtc.org
b92f6f9371
(Auto)update libjingle 71099685-> 71107853
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6674 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-14 18:22:37 +00:00
xians@webrtc.org
01bda2068b
Fixed the stats problem when new track is using the same ssrc as the previous track.
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Before this patch, when switching from voice mode to stereo mode, the stats won't be updated because StatsCollector binded the ssrc report with the old track, so the report can't be updated by the new track.
This patch fixes the porblem by changing the ssrc report track id to use the new track id.
TEST=libjingle_peerconnection_unittest --gtest_filter="*StatsCollectorTest*"
R=hta@chromium.org
Review URL: https://webrtc-codereview.appspot.com/17859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6632 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-09 07:38:38 +00:00
tommi@webrtc.org
ecb8723402
Change Timing::WallTimeNow to be static.
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There's no need to construct a Timing object to call this method.
On Windows we were unnecessarily calling CreateWaitableTimer + CloseHandle but never actually using that waitable timer.
There's otherwise no change in functionality.
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6624 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 12:48:29 +00:00
tkchin@webrtc.org
74bf7a6523
Add tkchin@ to OWNERS.
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Adding myself to OWNERS of subdirectories containing iOS bits. Added niklas.enbom@ for audio_device and wu@ for everything else.
R=niklas.enbom@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6578 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-01 18:38:28 +00:00
wu@webrtc.org
ec9f5fb34c
Change SdpSerializeCandidate to output candidate line without the "a=" and without the leading \r\n", i.e. candidate-attribute as defined in section 15.1 of [ICE].
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BUG=crbug/387632
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/17779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6533 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-24 17:05:10 +00:00
buildbot@webrtc.org
71dffb76dc
(Auto)update libjingle 69648312-> 69830415
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6527 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-24 07:24:49 +00:00
jiayl@webrtc.org
b43c99de29
Limits the send and receive buffer by bytes, not by packets.
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The new limit is 16MB for each buffer.
Also refactors the code to handle send failure more consistently.
BUG=3429
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21559005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6511 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-20 17:11:14 +00:00
jiayl@webrtc.org
db397e5c6c
Re-evalutes the ICE role on ICE restart.
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Also unifies the logic of ICE restart.
BUG=1775
R=juberti@google.com , juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6510 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-20 16:32:09 +00:00
buildbot@webrtc.org
bb2d65895b
(Auto)update libjingle 69617317-> 69623266
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6508 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-20 14:58:56 +00:00
buildbot@webrtc.org
58e7c8660c
(Auto)update libjingle 69588980-> 69589535
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6503 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-20 00:26:50 +00:00
buildbot@webrtc.org
fbd13286dc
(Auto)update libjingle 69555283-> 69567902
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6497 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-19 19:50:55 +00:00
buildbot@webrtc.org
d27d9ae644
(Auto)update libjingle 69506154-> 69515138
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6488 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-19 01:56:46 +00:00
jiayl@webrtc.org
f8063d34de
Properly shut down the SCTP stack.
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TBR phoglund@webrtc.org for the tsan_v2/suppressions.txt change.
R=ldixon@webrtc.org , pthatcher@webrtc.org
TBR=phoglund@webrtc.org
BUG=2749
Review URL: https://webrtc-codereview.appspot.com/12739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6484 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-18 21:30:40 +00:00
jiayl@webrtc.org
2eaac188bb
Makes the sid of a closed DataChannel available to reuse per the spec.
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BUG=2646
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6468 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 16:02:46 +00:00
buildbot@webrtc.org
44a317a698
(Auto)update libjingle 69337301-> 69359922
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6457 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 07:49:15 +00:00
buildbot@webrtc.org
27626a6256
(Auto)update libjingle 69278008-> 69291002
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6448 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16 13:39:40 +00:00
glaznev@webrtc.org
ab23d493e0
Add glaznev@ to OWNERS for webrtc/modules/video_capture and talk/app/webrtc.
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Review URL: https://webrtc-codereview.appspot.com/20659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6436 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-13 23:31:35 +00:00
glaznev@webrtc.org
c6c1dfd7ea
Add extra logging and latency restriction to VP8 HW encoder.
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- Do not allow encoder to accumulate more than 100 ms of
data in input buffers.
- Add optional extra logging (disabled by default) to track
encoder buffers timing.
BUG=
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6435 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-13 22:59:08 +00:00
buildbot@webrtc.org
a6764ab869
(Auto)update libjingle 69144530-> 69164179
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6434 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-13 18:24:39 +00:00
buildbot@webrtc.org
db56390f7e
(Auto)update libjingle 69143161-> 69144530
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6432 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-13 13:05:48 +00:00
buildbot@webrtc.org
c800c1cc40
(Auto)update libjingle 69131548-> 69132244
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6426 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-13 07:56:17 +00:00
buildbot@webrtc.org
7e71b77f8a
(Auto)update libjingle 69102234-> 69116997
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6424 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-13 01:14:01 +00:00
jiayl@webrtc.org
1a6c6281ca
Revert r6420 'Revert r6390 "Adds end to end DataChannel tests." Flaky on linux_memcheck'
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Failing tests are disabled for memcheck.
TBR=wu@webrtc.org
BUG=2626
Review URL: https://webrtc-codereview.appspot.com/13699004
Review URL: https://webrtc-codereview.appspot.com/13699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6422 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-12 21:59:29 +00:00
jiayl@webrtc.org
ddeec048c0
Revert r6390 "Adds end to end DataChannel tests." Flaky on linux_memcheck
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This reverts commit c3272a942f04f9dd0db3f6bf0d201bcf47c3fa08.
TBR=wu@webrtc.org
BUG=2626
Review URL: https://webrtc-codereview.appspot.com/13689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6420 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-12 21:42:46 +00:00