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
e61b8e32d8
Adds end to end DataChannel tests.
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BUG=2626
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6390 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-10 23:54:13 +00:00
94454b71ad
Fix the chain that propagates the audio frame's rtp and ntp timestamp including:
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* In AudioCodingModuleImpl::PlayoutData10Ms, don't reset the timestamp got from GetAudio.
* When there're more than one participant, set AudioFrame's RTP timestamp to 0.
* Copy ntp_time_ms_ in AudioFrame::CopyFrom method.
* In RemixAndResample, pass src frame's timestamp_ and ntp_time_ms_ to the dst frame.
* Fix how |elapsed_time_ms| is computed in channel.cc by adding GetPlayoutFrequency.
Tweaks on ntp_time_ms_:
* Init ntp_time_ms_ to -1 in AudioFrame ctor.
* When there're more than one participant, set AudioFrame's ntp_time_ms_ to an invalid value. I.e. we don't support ntp_time_ms_ in multiple participants case before the mixing is moved to chrome.
Added elapsed_time_ms to AudioFrame and pass it to chrome, where we don't have the information about the rtp timestmp's sample rate, i.e. can't convert rtp timestamp to ms.
BUG=3111
R=henrik.lundin@webrtc.org , turaj@webrtc.org , xians@webrtc.org
TBR=andrew
andrew to take another look on audio_conference_mixer_impl.cc
Review URL: https://webrtc-codereview.appspot.com/14559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6346 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-05 20:34:08 +00:00
5dc51fbe50
Closes the DataChannel when the send buffer is full or on transport errors.
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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
cb711f77d2
Add interface to propagate audio capture timestamp to the renderer.
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BUG=3111
R=andrew@webrtc.org , turaj@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6189 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-19 17:39:11 +00:00
3e01e0b16c
(Auto)update libjingle 66867790-> 66887616
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6128 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 17:54:10 +00:00
61c1b8ea32
(Auto)update libjingle 64585415-> 64594651
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@5870 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-09 06:06:38 +00:00
40b3b68cdf
Update libjingle 62364298->62472237
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@5632 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-03 18:30:11 +00:00
704bf9ebec
(Auto)update libjingle 62063505-> 62278774
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@5617 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-27 17:52:04 +00:00
a7b981843f
Unrevert 5590 "description"(=(Auto)update libjingle 61834300->61901702).
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BUG=N/A
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5595 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-21 15:51:43 +00:00
ef2215110c
Revert 5590 "description"
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> description
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8949006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5593 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-21 10:31:29 +00:00
2643805a20
description
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@5590 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-20 22:32:53 +00:00
385857dfd4
Update talk to 61549749.
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TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5549 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-14 00:56:12 +00:00
a576faf82a
Enable SCTP and use OPENSSL on Anroid and NSS on other platforms.
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It includes unit test fixes to properly initialize SSL if DTLS or SSL random number generator is used in the tests.
The private key and certificate constant strings used in some tests are updated to be compatible with NSS.
A few potentially overflow type conversions caused compiling warning on Windows and they are fixed by importing and using Chromium's checked_cast, which aborts the program if overflow occurs.
It also fixes a leak in nssstreamadapter.cc by releasing the PRFileDesc* in StreamClose.
BUG=2253
R=fischman@webrtc.org , juberti@google.com , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4679005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5459 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-29 17:45:53 +00:00
5bc25c41fc
Update libjingle to 57692857
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R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5217 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 00:24:06 +00:00
364f204d16
Update talk to 56698267.
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TBR=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/4119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5143 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20 21:49:41 +00:00
a23f0ca4ba
Update talk to 56619788
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R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3839005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5120 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-13 22:48:52 +00:00
07a6fbe83d
Update talk to 56092586.
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R=jiayl@webrtc.org , mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5078 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-04 18:41:34 +00:00
cecfd1832d
Update talk to 55821645.
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TEST=try bots
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5053 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-30 05:18:12 +00:00
97077a3ab2
Update libjingle to 55618622.
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Update libyuv to r826.
TEST=try bots
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5038 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-25 21:18:33 +00:00
8804a29951
Add CriticalSection to fakeaudiocapturemodule to protect the variables which will be accessed from process_thread_ and the main thread.
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TEST=try bots
BUG=1205
R=henrike@webrtc.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5019 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-22 23:09:20 +00:00
19f27e6a24
Update talk to 54527154.
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TBR=wu
Review URL: https://webrtc-codereview.appspot.com/2389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4954 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-13 17:18:27 +00:00
967bfff54d
Update talk to 52534915.
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R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/2251004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4786 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-19 05:49:50 +00:00
822fbd8b68
Update talk to 50918584.
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Together with Stefan's http://review.webrtc.org/1960004/ .
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2048004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-15 23:38:54 +00:00
91053e7c5a
Update libjingle to 50654631.
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R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2000006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4519 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-10 07:18:04 +00:00
28e2075280
Adds trunk/talk folder of revision 359 from libjingles google code to
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trunk/talk
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 00:45:36 +00:00
