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Author SHA1 Message Date
houssainy@google.com
e9b7d03db6 HTTPS Server used instead of HTTP for loading the bots to avoid the media permission pop-up clicks every time running the test. 
This code review based on the running issue:
https://webrtc-codereview.appspot.com/24939004/

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7497 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 16:34:25 +00:00
buildbot@webrtc.org
fb5410a8b7 (Auto)update libjingle 78262388-> 78262615 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7496 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 15:45:17 +00:00
pbos@webrtc.org
eacc6e4657 Remove some disabled tests in WebRtcVideoEngine2. 
Removes some tests that shouldn't have to be implemented or have already
been through other tests.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/25929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7495 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 15:36:54 +00:00
kjellander@webrtc.org
82e430c316 Suppress libyuv uninitialized read in CopyRow_AVX 
BUG=libyuv:377
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7494 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 13:51:49 +00:00
pbos@webrtc.org
32452b20b8 Make ReconfigureVideoEncoder use current bitrate. 
Prevents bitrate drops when changing resolution etc.

R=stefan@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/24069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7493 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 12:15:24 +00:00
kjellander@webrtc.org
860ccc9407 Tighten up MSan blacklist.txt owners. 
To avoid people adding stuff to the blacklist unless
it's really valid to do so.

BUG=
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7492 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 11:20:07 +00:00
pbos@webrtc.org
3f8f5554a0 Disable TestVp8Impl.BaseUnitTest on MSan. 
MemorySanitizer uses generic (non-optimized) libvpx which is not bit
exact. This may be a bug in upstream libvpx, but we're forced to disable
it now as it blocks launching the MSan bot.

R=stefan@webrtc.org
TBR=marpan@webrtc.org
BUG=3904

Review URL: https://webrtc-codereview.appspot.com/24089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7491 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 10:30:30 +00:00
stefan@webrtc.org
76960d5f74 For FIR packet, payload length is zero, so SendToNetwork function is failing. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7490 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 09:47:14 +00:00
kjellander@webrtc.org
1d9af96c06 Roll chromium_revision de13cf4..28d1981 (299488:300483) 
Mainly to pick up https://codereview.chromium.org/656293004/
to fix some MSan issues.

Summary of changes (de13cf4..28d1981/DEPS):
* third_party/android_tools d2b8620..36bf7ac
* third_party/libyuv 455c66b..5a09c3e (1038:1130)
* third_party/usrsctp/usrsctplib a11b3c5..7accb99
* tools/gyp 1977:1990
* tools/swarming_client c28b74f..a57d7db

Clang updated 217949:218707 (git diff de13cf4..28d1981 tools/clang/scripts/update.sh)

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7489 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 06:43:29 +00:00
aluebs@webrtc.org
67cf1d742b Break out WebRtcNs_Windowing function in ns_core 
This is done in order to make the code more readible and maintainable.
This introduces only +1 and -1 errors.

BUG=webrtc:3811
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7488 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-21 22:35:40 +00:00
aluebs@webrtc.org
0e7099244c Break out WebRtcNs_Energy function in ns_core 
This is done in order to make the code more readible and maintainable.
This generates bit-exact output.

BUG=webrtc:3811
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7487 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-21 22:14:10 +00:00
aluebs@webrtc.org
7634c09406 Break out WebRtcNs_IFFT function in ns_core 
This is done in order to make the code more readible and maintainable.
This creates bit-exact output.

BUG=webrtc:3811
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7486 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-21 21:27:00 +00:00
buildbot@webrtc.org
a5c36b397a (Auto)update libjingle 78193292-> 78199328 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7485 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-21 20:44:16 +00:00
guoweis@webrtc.org
b6173abe59 Fix local address leakage when IceTransportsType is relay 
As part of implementing IceTransportsType constraint, we should hide the raddr which is the mapped address to prevent local address leakage.

BUG=1179
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7484 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-21 20:40:21 +00:00
aluebs@webrtc.org
333e2556ed Break out WebRtcNs_UpdateBuffer function in ns_core 
This is done in order to make the code more readible and maintainable.
It generates bit-exact output.

BUG=webrtc:3811
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7483 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-21 20:33:09 +00:00
buildbot@webrtc.org
1288cbb704 (Auto)update libjingle 78106439-> 78193292 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7482 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-21 19:29:16 +00:00
henrik.lundin@webrtc.org
def1e97ed2 Implement AudioEncoderPcmU/A classes and convert AudioDecoder tests 
BUG=3926
R=kjellander@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7481 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-21 12:48:29 +00:00
bjornv@webrtc.org
78ea06dd34 audio_coding/codecs/ilbc: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> 
Removed usage of trivial macro.

BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7480 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-21 07:17:24 +00:00
henrik.lundin@webrtc.org
913f7b8d5e Fix for glitches in ACM when switching desired output sample rate 
The problem was that if the output sample rate is changed such from one
where no resampling is needed to a rate that requires resampling, the
first output from the resampler will contain an onset period. The
solution provided in this CL is to keep a copy of the last output frame
in ACM, and if the resampler is engaged, it will be primed with this
old frame before resampling the current frame.

BUG=3919
R=bjornv@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7479 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-21 06:54:23 +00:00
glaznev@webrtc.org
a8c0edd29f Avoid using EGLContext class for Android 4.1 and below. 
Support for this class was added in Android 4.2, so
disable surface decoding for lower Android versions.

BUG=3901
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7478 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-20 19:08:05 +00:00
bjornv@webrtc.org
b69ea9a35a common_audio: Replaced invalid operand in min_max_operations_neon.S" 
Vector Move immediate can not load #0x7FFF. Changed to us vdup from already loaded register.

BUG=N/A
TESTED=ios and android trybots
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7477 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-20 14:08:35 +00:00
pbos@webrtc.org
fa553ef605 Set up start bitrate in WebRtcVideoEngine2. 
R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/27789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7476 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-20 11:07:07 +00:00
pbos@webrtc.org
b35b136480 Make avg_{psnr,ssim}_threshold_ const. 
Triggered warning on next clang version being rolled as these variables
are annotated to be protected by crit_.

R=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/24949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7475 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-20 09:14:38 +00:00
bjornv@webrtc.org
2abebe7baf audio_coding/codecs/isac/main: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> 
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7474 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-20 08:26:41 +00:00
bjornv@webrtc.org
a5ce7bbe17 audio_coding/neteq: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> 
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7473 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-20 08:24:54 +00:00
henrike@webrtc.org
28100cb388 Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p." 
BUG=N/A
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7472 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 22:03:39 +00:00
buildbot@webrtc.org
7992b40994 (Auto)update libjingle 77953038-> 77970462 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7471 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 21:20:28 +00:00
henrike@webrtc.org
b1dac33cac Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..." 
BUG=3932
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/27779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7470 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 18:54:46 +00:00
glaznev@webrtc.org
58202946a7 Cleaning up Android AppRTCDemo. 
- Move signaling code from Activity to a separate class
and add interface for AppRTC signaling. For now
only pure GAE signaling implements this interface.
- Move peer connection, video source and peer connection
and SDP observer code from Activity to a separate class.
- Main Activity class will do only high level calls and
event handling for peer connection and signaling classes.
- Also add video renderer position update and use full
screen for local preview until the connection is established.

BUG=
R=braveyao@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7469 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 17:42:38 +00:00
houssainy@google.com
0371a37f85 Moving creating TURN configration to the host machine instead of the bots - rtcBot 
R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7468 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 16:43:50 +00:00
glaznev@webrtc.org
f7030d4ed7 Query Android device orientation on every camera frame received. 
Remove orientation listener from Android camera, since device
orientation change events are not well synchronized with actual
device display orientation. Plus these event may not be delivered
at all if device is in stationary position causing initial camera
frames appear rotated.

BUG=
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7467 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 16:25:06 +00:00
henrike@webrtc.org
9c58ea8d56 rtc_unittest: copied gtest excludes from libjingle_p2p_unittest since its tests have move to rtc_unittests. 
BUG=3925
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/28739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7466 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 16:12:33 +00:00
houssainy@google.com
c221db6165 Test names changed from e.g) testOneWayVideo/chrome=>chrome to testOneWayVideo/chrome-chrome. 
Because the symbol ">"  is interpreted as special command for output to file in bash commands.

TBR= andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7465 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 09:13:43 +00:00
henrik.lundin@webrtc.org
264e66f7a5 Add encoded_timestamp to AudioEncoder base class 
BUG=3926
TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7464 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-16 21:16:07 +00:00
henrik.lundin@webrtc.org
9ea6f8a84d New interface class AudioEncoder 
This class will be the base for new C++ wrapper classes for all
encoders.

BUG=3926
TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7463 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-16 11:26:24 +00:00
stefan@webrtc.org
8efaa270d8 Disable a bunch of Nat and Ice tests when running under DrMemory. 
BUG=3925
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7462 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-16 11:21:42 +00:00
andresp@webrtc.org
458c2c3b06 Improve rtcbot to load all test files at start and allow them to registerTests 
via: registerBotTest. After loading all tests main.js starts running the
requested one on the command arguments.

R=houssainy@google.com

Review URL: https://webrtc-codereview.appspot.com/29779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7461 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-16 07:36:37 +00:00
asapersson@webrtc.org
9aed002090 Add ability to include a larger time span (in addition to encode time) for measuring the processing time of a frame. 
Controlled by setting enable_extended_processing_usage. Enabled by default.

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7460 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-16 06:57:12 +00:00
henrike@webrtc.org
d1ba6d9cbf Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p. 
BUG=3379
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27709005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7459 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 17:30:28 +00:00
houssainy@google.com
3e2f8ff36c Selecting bot_type changed to be specified in the test file 
Selecting bot_type changed to be specified in the test file instead of
specify it in the running command.

Now we can write test for rtcBot that run one bot on chrome for android
and the other bot on chrome for desktop.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7458 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 15:01:11 +00:00
pbos@webrtc.org
e93cbd13d5 Fix data races in ThreadTest.ThreeThreadsInvoke. 
R=henrike@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/26819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7457 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 14:54:56 +00:00
bjornv@webrtc.org
f87c0aff7f audio_processing: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> 
Also includes a typo in a comment.
Affects
* aecm
* hpf

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7456 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 12:51:23 +00:00
bjornv@webrtc.org
f02ba9be54 audio_processing/agc: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> 
Affects AGC only.

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7455 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 11:16:48 +00:00
bjornv@webrtc.org
8dc00d76af audio_processing/ns: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> 
Affects fixed point version of Noise Suppression.

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7454 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 09:31:40 +00:00
henrik.lundin@webrtc.org
99e561f6a6 Extend AcmSwitchingOutputFrequencyOldApi with more frequencies 
Also reducing test duration, since the issue is triggered anyway.
The tests that are not failing are now enabled.

BUG=3919
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7453 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 08:50:00 +00:00
kjellander@webrtc.org
64f5611b3d Roll chromium_revision 2d714fa..de13cf4 (298667:299488) 
Mainly to pick up https://codereview.chromium.org/648613007
to fix some MSan issues.

Summary of changes (2d714fa..de13cf4/DEPS):
* third_party/boringssl 51fcd87..7ea8481
* third_party/icu d2abf6c..8ac906f
* third_party/usrsctp/usrsctplib dfd687b..a11b3c5

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7452 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 05:59:42 +00:00
bjornv@webrtc.org
fab5439112 common_audio: Removed version API from signal_processing 
The Signal Processing version API is not used anymore.

BUG=3353
R=kwiberg@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7451 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 04:38:42 +00:00
buildbot@webrtc.org
81ddc78536 (Auto)update libjingle 77701902-> 77709729 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7450 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 22:39:24 +00:00
buildbot@webrtc.org
1ecbe45c7e (Auto)update libjingle 77689511-> 77696841 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7449 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 20:29:28 +00:00
pbos@webrtc.org
43336b6b9f Remove unused (no-op) VideoOptions. 
Removing VideoOptions: adapt_input_to_encoder, adapt_view_switch,
video_one_layer_screencast and video_high_bitrate.

R=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/23079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7448 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 19:12:06 +00:00