asapersson@webrtc.org
580d367b14
Add macros and APIs for webrtc histograms.
...
BUG=crbug/419657
Code that links system_wrappers.gyp:system_wrappers should either:
- provide implementations for the APIs, or
- link with default implementations in system_wrappers.gyp:system_wrappers_default.
R=andresp@webrtc.org , kjellander@webrtc.org , mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7508 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 12:57:56 +00:00
buildbot@webrtc.org
9d446f2e16
(Auto)update libjingle 78296920-> 78342456
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7507 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 12:22:06 +00:00
kjellander@webrtc.org
8539bd0184
Download full Chromium checkouts by default
...
This changes sync_chromium.py to download a full Chromium
checkout instead of one with no history. It has been noticed
that the download of the no-history checkout is very slow, even
when on high-speed internet connections, due to current limitations
in the Git backend serving these clones.
Switching to a full checkout is faster, but requires more bandwidth
and disk space.
To keep the old behavior, users must set the CHROMIUM_NO_HISTORY
environment variable to 1.
Using a full checkout also enables the use of the Chromium
infrastructure teams' Git cache functionality, that speeds up
the initial download and also heavily reduces the traffic when
setting up multiple checkouts on the same machine.
This is not enabled by default, but is supported if the user is
setting the cache_dir variable in his checkout's .gclient file to
point at a directory on local disk.
BUG=3882
TESTED=
* Ran gclient sync and verified chromium/src now contained a Git
repo with full history.
* Tested rolling chromium_revision in DEPS forward + sync.
* Tested rolling it back again + sync.
* Tested with an existing no-history checkout:
CHROMIUM_NO_HISTORY=1 gclient sync
No change was performed.
* Tested with a .gclient that had cache_dir configured.
* Verified error message is displayed when .gclient has cache_dir
configured and CHROMIUM_NO_HISTORY=1.
R=iannucci@chromium.org
Review URL: https://webrtc-codereview.appspot.com/22869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7506 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 12:17:58 +00:00
stefan@webrtc.org
82462aade0
Adds support for sending first set of packets at increasingly higher bitrates to probe the link and faster ramp up to a high bitrate.
...
Also wires up a finch experiment to control this.
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7505 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 11:57:05 +00:00
houssainy@google.com
2192701135
Using the Unused turn configuration in two way test
...
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7504 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 08:40:53 +00:00
pbos@webrtc.org
ad553a2731
Let video_loopback use internal VCM capturers.
...
R=stefan@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/27819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7503 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 08:24:02 +00:00
andrew@webrtc.org
15c717beee
Add a memcheck exclusion for EndToEndTest.CanSwitchToUseAllSsrcs.
...
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7502 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 05:37:37 +00:00
buildbot@webrtc.org
a9f0898e7d
(Auto)update libjingle 78273470-> 78296920
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7501 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-22 22:02:00 +00:00
glaznev@webrtc.org
7bb4a9881d
Merging Henrik's and Peter's changes for AppRTCDemo
...
from https://github.com/hkjellander/AppRTCDemo .
Description of changes:
- Add connect screen with an option to enter room number or select loopback mode.
- Add 'hangup' and 'WebRTC statistics' buttons to AppRTCDemo activity.
BUG=3938
R=kjellander@webrtc.org , pbos@webrtc.org , pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7500 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-22 17:43:37 +00:00
houssainy@google.com
fce8f5d319
NOTE: 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/29819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7499 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-22 17:24:20 +00:00
houssainy@google.com
3382059e55
Adding Two way video and audio streaming test to RtcBot
...
NOTE: This code review based on this running issue:
https://review.webrtc.org/24939004/
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7498 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-22 17:17:15 +00:00
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.
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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
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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.
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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.
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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