jiayl@webrtc.org
6c6f33b5bb
Fix the flaky RTP DataChannel test.
...
BUG=2891
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6418 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-12 21:05:19 +00:00
buildbot@webrtc.org
18dfa8d574
(Auto)update libjingle 69069003-> 69082899
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6417 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-12 18:11:02 +00:00
xians@webrtc.org
4cb012858f
Fixed GetStats when local and remote track are using the same ssrc.
...
R=hta@chromium.org , kjellander@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6414 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-12 14:57:05 +00:00
buildbot@webrtc.org
b90619c07f
(Auto)update libjingle 69049090-> 69054765
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6412 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-12 09:19:08 +00:00
buildbot@webrtc.org
d41eaeb7cd
(Auto)update libjingle 69005149-> 69049090
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6408 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-12 07:13:26 +00:00
buildbot@webrtc.org
e9e8007ab4
(Auto)update libjingle 68985065-> 69005149
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6406 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-11 18:41:17 +00:00
pbos@webrtc.org
9e65a3b013
Re-land webrtcmediaengine.cc part of r6397.
...
webrtcvideoengine.cc un-reverted by a bot roll in r6399 so half of r6397
is still applied. The applied fix (diff between r6397) is to put
WebRtcVideoEngine2 in ifdefs and only build for WEBRTC_CHROMIUM_BUILDs
corresponding to webrtcmediaengine.h.
BUG=
R=minyue@webrtc.org
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19719005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6401 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-11 13:42:37 +00:00
buildbot@webrtc.org
5d223a7d2d
(Auto)update libjingle 68982444-> 68983526
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6399 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-11 13:05:05 +00:00
minyue@webrtc.org
6604c6df26
Revert 6397 "(Auto)update libjingle 68949184-> 68982444"
...
> (Auto)update libjingle 68949184-> 68982444
TBR=buildbot@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6398 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-11 13:02:36 +00:00
buildbot@webrtc.org
af214d804f
(Auto)update libjingle 68949184-> 68982444
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6397 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-11 12:46:49 +00:00
jiayl@webrtc.org
e61b8e32d8
Adds end to end DataChannel tests.
...
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
glaznev@webrtc.org
a40210aee2
Add support for NVidia VP8 HW encoder.
...
- Some changes in HW VP8 encoder search logic to detect HW codec
with supported color space format.
- Support yuv420 and nv12 formants for encoder input.
- Add some extra logging and encoder frame drop statistics.
BUG=3176
R=fischman@webrtc.org , tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6389 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-10 23:48:29 +00:00
kjellander@webrtc.org
1014101470
Revert 6380 "Replace libjingle_root with talk_root variable."
...
It turns out this doesn't fix the problem we're trying to solve...
> Replace libjingle_root with talk_root variable.
>
> This CL is similar to https://review.webrtc.org/9019004/
> It is needed in order to be able to build with different
> copies of libjingle. Having the libjingle_root variable didn't
> make this possible, since relative paths in the .isolate files
> ended up at the wrong directory level and .isolate files doesn't
> support all the normal GYP variables like <(DEPTH).
>
> BUG=chromium:343106
> TEST=trybots passing compile step with clobber.
> R=tommi@webrtc.org , wu@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/15709004
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6384 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-10 10:13:38 +00:00
buildbot@webrtc.org
3eb2c2f4c3
(Auto)update libjingle 68891947-> 68893961
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6383 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-10 09:39:06 +00:00
pbos@webrtc.org
86f613d6b8
Move WebRtcVideoEngine2 fakes to unittest header.
...
BUG=1788
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6382 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-10 08:53:05 +00:00
kjellander@webrtc.org
0238682984
Replace libjingle_root with talk_root variable.
...
This CL is similar to https://review.webrtc.org/9019004/
It is needed in order to be able to build with different
copies of libjingle. Having the libjingle_root variable didn't
make this possible, since relative paths in the .isolate files
ended up at the wrong directory level and .isolate files doesn't
support all the normal GYP variables like <(DEPTH).
BUG=chromium:343106
TEST=trybots passing compile step with clobber.
R=tommi@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6380 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-10 05:46:31 +00:00
kjellander@webrtc.org
6b6e58d632
Remove unused test_env.py from isolate files + fix nss path.
...
This is not necessary for executing tests for WebRTC.
It probably appeared in our .isolate files because of code
copied from Chromium.
BUG=
TEST=All non-baremetal trybots passing.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6373 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-09 14:35:09 +00:00
stefan@webrtc.org
85d2794e5b
Adds support for the "apt" format parameter and turns on the RTX feature.
...
BUG=1811,1095
R=henrike@webrtc.org , mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12579009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6372 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-09 12:51:39 +00:00
jiayl@webrtc.org
e3cdd9959e
Revert "Fix the "Failed unprotect audio RTP packet" error when SCTP is bundled with audio."
...
This reverts commit 56631a14bdae24aa0bfaceeb2b57df729fee1227.
TBR=henrike@webrtc.org
BUG=3235
Review URL: https://webrtc-codereview.appspot.com/19669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6363 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-06 22:32:57 +00:00
tkchin@webrtc.org
013bdf802a
APPRTCDemo(objc): Remove regex parsing in favor of JSON struct.
...
Also some cleanup/refactoring of APPRTCAppClient.
R=fischman@webrtc.org , glaznev@webrtc.org
BUG=3407
Review URL: https://webrtc-codereview.appspot.com/18499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6362 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-06 22:29:10 +00:00
glaznev@webrtc.org
c3288c130d
Add OpenGL Android video renderer which can display multiple
...
yuv420 images in a single GLSurfaceView.
Start using new video renderer in AppRTC demo app.
BUG=
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6360 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-06 21:57:46 +00:00
jiayl@webrtc.org
745a39cced
Fix the "Failed unprotect audio RTP packet" error when SCTP is bundled with audio.
...
BUG=3235
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6356 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-06 19:24:02 +00:00
fischman@webrtc.org
9512719569
AppRTCDemo(android): support app (UI) & capture rotation.
...
Now app UI rotates as the device orientation changes, and the captured stream
tries to maintain real-world-up, matching Chrome/Android and Hangouts/Android
behavior.
BUG=2432
R=glaznev@webrtc.org , henrike@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15689005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6354 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-06 18:40:44 +00:00
buildbot@webrtc.org
91c910469f
(Auto)update libjingle 68701339-> 68703656
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6352 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-06 16:29:00 +00:00
pbos@webrtc.org
910473b31a
Fix C++11 -Wnarrowing in channel_unittest.cc.
...
Implicit conversion from int to unsigned char inside {} initializers is
ill-formed C++11 and triggers a warning in clang when building it as
such.
BUG=
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6351 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-06 15:44:00 +00:00
buildbot@webrtc.org
7b6cbb3aa0
(Auto)update libjingle 68689052-> 68689059
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6350 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-06 10:54:08 +00:00
pbos@webrtc.org
6ae48c6609
Make VideoSendStream/VideoReceiveStream configs const.
...
Benefits of this is that the send config previously had unclear locking
requirements, a lock was used to lock parts parts of it while
reconfiguring the VideoEncoder. Primary work was splitting out video
streams from config as well as encoder_settings as these change on
ReconfigureVideoEncoder. Now threading requirements for both member
configs are clear (as they are read-only), and encoder_settings doesn't
stay in the config as a stale pointer.
CreateVideoSendStream now takes video streams separately as well as the
encoder_settings pointer, analogous to ReconfigureVideoEncoder.
This change required changing so that pacing is silently enabled when
using suspend_below_min_bitrate rather than silently setting it.
R=henrik.lundin@webrtc.org , mflodman@webrtc.org , pthatcher@webrtc.org , stefan@webrtc.org
BUG=3260
Review URL: https://webrtc-codereview.appspot.com/20409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6349 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-06 10:49:19 +00:00
buildbot@webrtc.org
4b83a471de
(Auto)update libjingle 68646004-> 68648993
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6348 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-05 21:11:28 +00:00
wu@webrtc.org
94454b71ad
Fix the chain that propagates the audio frame's rtp and ntp timestamp including:
...
* 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
fischman@webrtc.org
130fa64d4c
AppRTCDemo(android): remove HTML/regex hackery in favor of JSON struct.
...
BUG=3407
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16619006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6345 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-05 20:31:41 +00:00
pbos@webrtc.org
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
buildbot@webrtc.org
f1adbeedb4
(Auto)update libjingle 68562943-> 68571194
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6333 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-04 21:57:16 +00:00
tkchin@webrtc.org
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
buildbot@webrtc.org
6f237769b3
(Auto)update libjingle 68507189-> 68543735
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6329 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-04 16:23:10 +00:00
buildbot@webrtc.org
40b45fc07a
(Auto)update libjingle 68506654-> 68507189
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6328 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-04 14:48:33 +00:00
buildbot@webrtc.org
0cdcd23a03
(Auto)update libjingle 68501302-> 68506654
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6321 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-04 01:31:14 +00:00
buildbot@webrtc.org
af81b9bffd
(Auto)update libjingle 68499439-> 68501302
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6320 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-04 00:08:54 +00:00
buildbot@webrtc.org
251fdf64cb
(Auto)update libjingle 68495561-> 68499439
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6319 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-03 23:43:48 +00:00
henrike@webrtc.org
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
buildbot@webrtc.org
53217848b2
(Auto)update libjingle 68465410-> 68487517
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6317 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-03 21:09:11 +00:00
fischman@webrtc.org
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
pbos@webrtc.org
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
buildbot@webrtc.org
b525a9d790
(Auto)update libjingle 68379861-> 68445177
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6309 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-03 09:42:15 +00:00
pbos@webrtc.org
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
buildbot@webrtc.org
34a08b4fb8
(Auto)update libjingle 68275107-> 68379861
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6305 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-02 15:48:10 +00:00
pbos@webrtc.org
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
jiayl@webrtc.org
8a09af3f67
Fix the build error from OpenSSLStreamAdapter::SSLVerifyCallback
...
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
jiayl@webrtc.org
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
tkchin@webrtc.org
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
tkchin@webrtc.org
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