external/webrtc.git
20 min agoAdd HD support to Android if we detect a hardware video encoder that can be used... master
perkj@webrtc.org [Fri, 24 Oct 2014 11:38:19 +0000 (11:38 +0000)]
Add HD support to Android if we detect a hardware video encoder that can be used. This Change the internal class MediaCodecVideoEncoder to have a one public method for checking if the platform is supported. It also adds &hd=true to the reqest url a hardware encoder is detected.

BUG=3934
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7520 4adac7df-926f-26a2-2b94-8c16560cd09d

2 hours agoAdding the subtool rtcBot report visualizer
houssainy@google.com [Fri, 24 Oct 2014 09:26:16 +0000 (09:26 +0000)]
Adding the subtool rtcBot report visualizer

This tool for visualize the output reports of rtcBot by calculating
the average and max of a specific stats and plot the output.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7519 4adac7df-926f-26a2-2b94-8c16560cd09d

2 hours agoMove min transmit bitrate to VideoEncoderConfig.
pbos@webrtc.org [Fri, 24 Oct 2014 09:23:21 +0000 (09:23 +0000)]
Move min transmit bitrate to VideoEncoderConfig.

min_transmit_bitrate_bps needs to be reconfigurable during a call (since
this is currently set only for screensharing through libjingle and can't
be set once and for all for the entire Call.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/28779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7518 4adac7df-926f-26a2-2b94-8c16560cd09d

12 hours agopatch from issue 25469004
pthatcher@webrtc.org [Thu, 23 Oct 2014 23:37:22 +0000 (23:37 +0000)]
patch from issue 25469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7517 4adac7df-926f-26a2-2b94-8c16560cd09d

12 hours ago(Auto)update libjingle 78381351-> 78389679
buildbot@webrtc.org [Thu, 23 Oct 2014 23:07:23 +0000 (23:07 +0000)]
(Auto)update libjingle 78381351-> 78389679

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7516 4adac7df-926f-26a2-2b94-8c16560cd09d

14 hours ago(Auto)update libjingle 78344087-> 78381351
buildbot@webrtc.org [Thu, 23 Oct 2014 21:36:17 +0000 (21:36 +0000)]
(Auto)update libjingle 78344087-> 78381351

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7515 4adac7df-926f-26a2-2b94-8c16560cd09d

16 hours agoBreak out WebRtcNs_ComputeDdUpdate function in ns_core
aluebs@webrtc.org [Thu, 23 Oct 2014 19:54:33 +0000 (19:54 +0000)]
Break out WebRtcNs_ComputeDdUpdate 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/31739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7514 4adac7df-926f-26a2-2b94-8c16560cd09d

16 hours agoBreak out WebRtcNs_UpdateNoise function in ns_core
aluebs@webrtc.org [Thu, 23 Oct 2014 19:49:42 +0000 (19:49 +0000)]
Break out WebRtcNs_UpdateNoise 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

Review URL: https://webrtc-codereview.appspot.com/25889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7513 4adac7df-926f-26a2-2b94-8c16560cd09d

16 hours agoBreak out FFT function in ns_core
aluebs@webrtc.org [Thu, 23 Oct 2014 19:36:42 +0000 (19:36 +0000)]
Break out FFT function in ns_core

This is done in order to make the code more readible and maintainable.
This introduces an error of only +1 and -1.

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

Review URL: https://webrtc-codereview.appspot.com/27749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7512 4adac7df-926f-26a2-2b94-8c16560cd09d

16 hours agoBreak out ComputeSnr function in ns_core
aluebs@webrtc.org [Thu, 23 Oct 2014 19:34:14 +0000 (19:34 +0000)]
Break out ComputeSnr function in ns_core

This is done in order to make the code more readible and maintainable.
The output is bit-exact.

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

Review URL: https://webrtc-codereview.appspot.com/31729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7511 4adac7df-926f-26a2-2b94-8c16560cd09d

19 hours agoAdding three video conference bots test
houssainy@google.com [Thu, 23 Oct 2014 16:45:07 +0000 (16:45 +0000)]
Adding three video conference bots test

A video conference between three bots, each bot creating two
peerConnections, and each peer connected to one of the other bots.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7510 4adac7df-926f-26a2-2b94-8c16560cd09d

20 hours agoAdding file from test.webrtc.org domain to be downloaded
houssainy@google.com [Thu, 23 Oct 2014 15:41:30 +0000 (15:41 +0000)]
Adding file from test.webrtc.org domain to be downloaded

This has been configured to allow cross domain to access this generated
file:
https://test.webrtc.org/test-download-file/9000KB.data

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7509 4adac7df-926f-26a2-2b94-8c16560cd09d

23 hours agoAdd macros and APIs for webrtc histograms.
asapersson@webrtc.org [Thu, 23 Oct 2014 12:57:56 +0000 (12:57 +0000)]
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

23 hours ago(Auto)update libjingle 78296920-> 78342456
buildbot@webrtc.org [Thu, 23 Oct 2014 12:22:06 +0000 (12:22 +0000)]
(Auto)update libjingle 78296920-> 78342456

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7507 4adac7df-926f-26a2-2b94-8c16560cd09d

23 hours agoDownload full Chromium checkouts by default
kjellander@webrtc.org [Thu, 23 Oct 2014 12:17:58 +0000 (12:17 +0000)]
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

24 hours agoAdds support for sending first set of packets at increasingly higher bitrates to...
stefan@webrtc.org [Thu, 23 Oct 2014 11:57:05 +0000 (11:57 +0000)]
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

27 hours agoUsing the Unused turn configuration in two way test
houssainy@google.com [Thu, 23 Oct 2014 08:40:53 +0000 (08:40 +0000)]
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

27 hours agoLet video_loopback use internal VCM capturers.
pbos@webrtc.org [Thu, 23 Oct 2014 08:24:02 +0000 (08:24 +0000)]
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

30 hours agoAdd a memcheck exclusion for EndToEndTest.CanSwitchToUseAllSsrcs.
andrew@webrtc.org [Thu, 23 Oct 2014 05:37:37 +0000 (05:37 +0000)]
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

37 hours ago(Auto)update libjingle 78273470-> 78296920
buildbot@webrtc.org [Wed, 22 Oct 2014 22:02:00 +0000 (22:02 +0000)]
(Auto)update libjingle 78273470-> 78296920

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7501 4adac7df-926f-26a2-2b94-8c16560cd09d

42 hours agoMerging Henrik's and Peter's changes for AppRTCDemo
glaznev@webrtc.org [Wed, 22 Oct 2014 17:43:37 +0000 (17:43 +0000)]
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

42 hours agoNOTE: This code review based on the running issue:
houssainy@google.com [Wed, 22 Oct 2014 17:24:20 +0000 (17:24 +0000)]
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

42 hours agoAdding Two way video and audio streaming test to RtcBot
houssainy@google.com [Wed, 22 Oct 2014 17:17:15 +0000 (17:17 +0000)]
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

43 hours agoHTTPS Server used instead of HTTP for loading the bots to avoid the media permission...
houssainy@google.com [Wed, 22 Oct 2014 16:34:25 +0000 (16:34 +0000)]
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

44 hours ago(Auto)update libjingle 78262388-> 78262615
buildbot@webrtc.org [Wed, 22 Oct 2014 15:45:17 +0000 (15:45 +0000)]
(Auto)update libjingle 78262388-> 78262615

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7496 4adac7df-926f-26a2-2b94-8c16560cd09d

44 hours agoRemove some disabled tests in WebRtcVideoEngine2.
pbos@webrtc.org [Wed, 22 Oct 2014 15:36:54 +0000 (15:36 +0000)]
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

46 hours agoSuppress libyuv uninitialized read in CopyRow_AVX
kjellander@webrtc.org [Wed, 22 Oct 2014 13:51:49 +0000 (13:51 +0000)]
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

47 hours agoMake ReconfigureVideoEncoder use current bitrate.
pbos@webrtc.org [Wed, 22 Oct 2014 12:15:24 +0000 (12:15 +0000)]
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

2 days agoTighten up MSan blacklist.txt owners.
kjellander@webrtc.org [Wed, 22 Oct 2014 11:20:07 +0000 (11:20 +0000)]
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

2 days agoDisable TestVp8Impl.BaseUnitTest on MSan.
pbos@webrtc.org [Wed, 22 Oct 2014 10:30:30 +0000 (10:30 +0000)]
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

2 days agoFor FIR packet, payload length is zero, so SendToNetwork function is failing.
stefan@webrtc.org [Wed, 22 Oct 2014 09:47:14 +0000 (09:47 +0000)]
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

2 days agoRoll chromium_revision de13cf4..28d1981 (299488:300483)
kjellander@webrtc.org [Wed, 22 Oct 2014 06:43:29 +0000 (06:43 +0000)]
Roll chromium_revision de13cf4..28d1981 (299488:300483)

Mainly to pick up https://codereview.chromium.org/656293004/
to fix some MSan issues.

Summary of changes (https://chromium.googlesource.com/chromium/src/+/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

2 days agoBreak out WebRtcNs_Windowing function in ns_core
aluebs@webrtc.org [Tue, 21 Oct 2014 22:35:40 +0000 (22:35 +0000)]
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

2 days agoBreak out WebRtcNs_Energy function in ns_core
aluebs@webrtc.org [Tue, 21 Oct 2014 22:14:10 +0000 (22:14 +0000)]
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

2 days agoBreak out WebRtcNs_IFFT function in ns_core
aluebs@webrtc.org [Tue, 21 Oct 2014 21:27:00 +0000 (21:27 +0000)]
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

2 days ago(Auto)update libjingle 78193292-> 78199328
buildbot@webrtc.org [Tue, 21 Oct 2014 20:44:16 +0000 (20:44 +0000)]
(Auto)update libjingle 78193292-> 78199328

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7485 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoFix local address leakage when IceTransportsType is relay
guoweis@webrtc.org [Tue, 21 Oct 2014 20:40:21 +0000 (20:40 +0000)]
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

2 days agoBreak out WebRtcNs_UpdateBuffer function in ns_core
aluebs@webrtc.org [Tue, 21 Oct 2014 20:33:09 +0000 (20:33 +0000)]
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

2 days ago(Auto)update libjingle 78106439-> 78193292
buildbot@webrtc.org [Tue, 21 Oct 2014 19:29:16 +0000 (19:29 +0000)]
(Auto)update libjingle 78106439-> 78193292

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7482 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoImplement AudioEncoderPcmU/A classes and convert AudioDecoder tests
henrik.lundin@webrtc.org [Tue, 21 Oct 2014 12:48:29 +0000 (12:48 +0000)]
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

3 days agoaudio_coding/codecs/ilbc: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>
bjornv@webrtc.org [Tue, 21 Oct 2014 07:17:24 +0000 (07:17 +0000)]
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

3 days agoFix for glitches in ACM when switching desired output sample rate
henrik.lundin@webrtc.org [Tue, 21 Oct 2014 06:54:23 +0000 (06:54 +0000)]
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

3 days agoAvoid using EGLContext class for Android 4.1 and below.
glaznev@webrtc.org [Mon, 20 Oct 2014 19:08:05 +0000 (19:08 +0000)]
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

3 days agocommon_audio: Replaced invalid operand in min_max_operations_neon.S"
bjornv@webrtc.org [Mon, 20 Oct 2014 14:08:35 +0000 (14:08 +0000)]
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

4 days agoSet up start bitrate in WebRtcVideoEngine2.
pbos@webrtc.org [Mon, 20 Oct 2014 11:07:07 +0000 (11:07 +0000)]
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

4 days agoMake avg_{psnr,ssim}_threshold_ const.
pbos@webrtc.org [Mon, 20 Oct 2014 09:14:38 +0000 (09:14 +0000)]
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

4 days agoaudio_coding/codecs/isac/main: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>
bjornv@webrtc.org [Mon, 20 Oct 2014 08:26:41 +0000 (08:26 +0000)]
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

4 days agoaudio_coding/neteq: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>
bjornv@webrtc.org [Mon, 20 Oct 2014 08:24:54 +0000 (08:24 +0000)]
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

6 days agoReverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp...
henrike@webrtc.org [Fri, 17 Oct 2014 22:03:39 +0000 (22:03 +0000)]
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

6 days ago(Auto)update libjingle 77953038-> 77970462
buildbot@webrtc.org [Fri, 17 Oct 2014 21:20:28 +0000 (21:20 +0000)]
(Auto)update libjingle 77953038-> 77970462

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7471 4adac7df-926f-26a2-2b94-8c16560cd09d

6 days agoRevert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..."
henrike@webrtc.org [Fri, 17 Oct 2014 18:54:46 +0000 (18:54 +0000)]
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

6 days agoCleaning up Android AppRTCDemo.
glaznev@webrtc.org [Fri, 17 Oct 2014 17:42:38 +0000 (17:42 +0000)]
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

6 days agoMoving creating TURN configration to the host machine instead of the bots - rtcBot
houssainy@google.com [Fri, 17 Oct 2014 16:43:50 +0000 (16:43 +0000)]
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

6 days agoQuery Android device orientation on every camera frame received.
glaznev@webrtc.org [Fri, 17 Oct 2014 16:25:06 +0000 (16:25 +0000)]
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

6 days agortc_unittest: copied gtest excludes from libjingle_p2p_unittest since its tests have...
henrike@webrtc.org [Fri, 17 Oct 2014 16:12:33 +0000 (16:12 +0000)]
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

7 days agoTest names changed from e.g) testOneWayVideo/chrome=>chrome to testOneWayVideo/chrome...
houssainy@google.com [Fri, 17 Oct 2014 09:13:43 +0000 (09:13 +0000)]
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

7 days agoAdd encoded_timestamp to AudioEncoder base class
henrik.lundin@webrtc.org [Thu, 16 Oct 2014 21:16:07 +0000 (21:16 +0000)]
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

8 days agoNew interface class AudioEncoder
henrik.lundin@webrtc.org [Thu, 16 Oct 2014 11:26:24 +0000 (11:26 +0000)]
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

8 days agoDisable a bunch of Nat and Ice tests when running under DrMemory.
stefan@webrtc.org [Thu, 16 Oct 2014 11:21:42 +0000 (11:21 +0000)]
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

8 days agoImprove rtcbot to load all test files at start and allow them to registerTests
andresp@webrtc.org [Thu, 16 Oct 2014 07:36:37 +0000 (07:36 +0000)]
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

8 days agoAdd ability to include a larger time span (in addition to encode time) for measuring...
asapersson@webrtc.org [Thu, 16 Oct 2014 06:57:12 +0000 (06:57 +0000)]
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

8 days agoCreate a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p.
henrike@webrtc.org [Wed, 15 Oct 2014 17:30:28 +0000 (17:30 +0000)]
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

8 days agoSelecting bot_type changed to be specified in the test file
houssainy@google.com [Wed, 15 Oct 2014 15:01:11 +0000 (15:01 +0000)]
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

8 days agoFix data races in ThreadTest.ThreeThreadsInvoke.
pbos@webrtc.org [Wed, 15 Oct 2014 14:54:56 +0000 (14:54 +0000)]
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

8 days agoaudio_processing: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>
bjornv@webrtc.org [Wed, 15 Oct 2014 12:51:23 +0000 (12:51 +0000)]
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

9 days agoaudio_processing/agc: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>
bjornv@webrtc.org [Wed, 15 Oct 2014 11:16:48 +0000 (11:16 +0000)]
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

9 days agoaudio_processing/ns: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>
bjornv@webrtc.org [Wed, 15 Oct 2014 09:31:40 +0000 (09:31 +0000)]
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

9 days agoExtend AcmSwitchingOutputFrequencyOldApi with more frequencies
henrik.lundin@webrtc.org [Wed, 15 Oct 2014 08:50:00 +0000 (08:50 +0000)]
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

9 days agoRoll chromium_revision 2d714fa..de13cf4 (298667:299488)
kjellander@webrtc.org [Wed, 15 Oct 2014 05:59:42 +0000 (05:59 +0000)]
Roll chromium_revision 2d714fa..de13cf4 (298667:299488)

Mainly to pick up https://codereview.chromium.org/648613007
to fix some MSan issues.

Summary of changes (https://chromium.googlesource.com/chromium/src/+/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

9 days agocommon_audio: Removed version API from signal_processing
bjornv@webrtc.org [Wed, 15 Oct 2014 04:38:42 +0000 (04:38 +0000)]
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

9 days ago(Auto)update libjingle 77701902-> 77709729
buildbot@webrtc.org [Tue, 14 Oct 2014 22:39:24 +0000 (22:39 +0000)]
(Auto)update libjingle 77701902-> 77709729

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7450 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days ago(Auto)update libjingle 77689511-> 77696841
buildbot@webrtc.org [Tue, 14 Oct 2014 20:29:28 +0000 (20:29 +0000)]
(Auto)update libjingle 77689511-> 77696841

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7449 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoRemove unused (no-op) VideoOptions.
pbos@webrtc.org [Tue, 14 Oct 2014 19:12:06 +0000 (19:12 +0000)]
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

9 days agolibjingle: use _stricmp instead of deprecated stricmp.
henrike@webrtc.org [Tue, 14 Oct 2014 17:07:41 +0000 (17:07 +0000)]
libjingle: use _stricmp instead of deprecated stricmp.

BUG=N/A
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7447 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days agoRemove -1 from Call::Config::start_bitrate_bps.
pbos@webrtc.org [Tue, 14 Oct 2014 11:52:10 +0000 (11:52 +0000)]
Remove -1 from Call::Config::start_bitrate_bps.

Instead initialize it to a good default value. The code does the same,
but we don't have to check explicitly for -1.

R=mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/23989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7445 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days agoAdd periodic logging of received RTP headers and estimated clock offsets for e2e...
stefan@webrtc.org [Tue, 14 Oct 2014 11:40:13 +0000 (11:40 +0000)]
Add periodic logging of received RTP headers and estimated clock offsets for e2e delay.

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7444 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days agoNew ACM test to trigger audio glitch when switching output sample rate
henrik.lundin@webrtc.org [Tue, 14 Oct 2014 10:49:58 +0000 (10:49 +0000)]
New ACM test to trigger audio glitch when switching output sample rate

This CL implements a new unit test. The test is designed to trigger
a problem in ACM where switching the desired output frequency creates
a short discontinuity in the output audio. The problem itself is not
solved in this CL, but the failing test is disabled for now.

BUG=3919
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7443 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days agoAdd a packet loss full stack test to the new API.
stefan@webrtc.org [Tue, 14 Oct 2014 10:38:49 +0000 (10:38 +0000)]
Add a packet loss full stack test to the new API.

Remove all full stack tests for the old API.

BUG=3750
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7442 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days agoWorkarounds for a bug in VS2013.3 linker when PGO is turned on.
kwiberg@webrtc.org [Tue, 14 Oct 2014 09:40:04 +0000 (09:40 +0000)]
Workarounds for a bug in VS2013.3 linker when PGO is turned on.

See crbug.com/421607 for more details about this. This CL solve a linker bug when the PGO is turned on, without changing the behaviour or the performances.

BUG=crbug.com/421607
R=kwiberg@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26789005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7441 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days agoWire up external encoders.
pbos@webrtc.org [Tue, 14 Oct 2014 04:25:33 +0000 (04:25 +0000)]
Wire up external encoders.

R=pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/30649005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7440 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days ago(Auto)update libjingle 77554188-> 77629208
buildbot@webrtc.org [Tue, 14 Oct 2014 01:17:42 +0000 (01:17 +0000)]
(Auto)update libjingle 77554188-> 77629208

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7439 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days agoMove exlusion of VP9 integration tests for DrMemory
marpan@webrtc.org [Tue, 14 Oct 2014 00:34:19 +0000 (00:34 +0000)]
Move exlusion of VP9 integration tests for DrMemory
from modules_unittests to modules_tests file.

Also rename and move ProcessNoLossChangeBitRate,
and move TestVp8Impl.BaseUnitTest to proper file.

The previous commit r7435 disabled it in the wrong file.

TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7438 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days agoAdjust speech probability in NS when echo
aluebs@webrtc.org [Mon, 13 Oct 2014 20:48:05 +0000 (20:48 +0000)]
Adjust speech probability in NS when echo

The average speech probability for the higher band is multiplied by the quotient of the process and analyze powers, to avoid thinking that suppressed echo is speech. In order to do this both magnitudes, alanyze and process, needed to be stored. This also was used to calculate different previous STSA estimates for analyze and process.
This CL was tested on two long team member recordings (bjornv and kwiberg) and the noisiest (5) recordings from the QA set.

BUG=webrtc:3763
R=andrew@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7437 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days agoRemoves xmllite from talk/xmllite since webrtc/xmllite is used instead.
henrike@webrtc.org [Mon, 13 Oct 2014 18:27:11 +0000 (18:27 +0000)]
Removes xmllite from talk/xmllite since webrtc/xmllite is used instead.

BUG=3379
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/23039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7436 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days agoDisable VP9 integration tests on DrMemory.
marpan@webrtc.org [Mon, 13 Oct 2014 17:10:40 +0000 (17:10 +0000)]
Disable VP9 integration tests on DrMemory.

Will try re-enabling them on next libvpx roll using faster codec speed setting.

BUG=3917

TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7435 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days agocommon_audio: Removed macro WEBRTC_SPL_RSHIFT_W16
bjornv@webrtc.org [Mon, 13 Oct 2014 14:00:43 +0000 (14:00 +0000)]
common_audio: Removed macro WEBRTC_SPL_RSHIFT_W16

Replaced the trivial right shift macro at remaining 4 places and removed from signal_processing.

Affected components:
* vad
* aecm

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

Review URL: https://webrtc-codereview.appspot.com/25849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7434 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days agoiSAC tests: Type buffers as uint8_t[] to avoid casts
kwiberg@webrtc.org [Mon, 13 Oct 2014 13:29:04 +0000 (13:29 +0000)]
iSAC tests: Type buffers as uint8_t[] to avoid casts

The iSAC interface functions now expect uint8_t arrays, so change some
arrays to be of that type instead of casting at each point of use.

R=bjornv@webrtc.org, henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7433 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days agoaudio_processing: Replaced macro WEBRTC_SPL_RSHIFT_W16 with >>
bjornv@webrtc.org [Mon, 13 Oct 2014 13:01:13 +0000 (13:01 +0000)]
audio_processing: Replaced macro WEBRTC_SPL_RSHIFT_W16 with >>

The implementation of WEBRTC_SPL_RSHIFT_W16 is simply >>. This CL removes the macro usage in audio_processing and signal_processing.

Affected components:
* aecm
* agc
* nsx

Indirectly affecting (through signal_processing changes)
* codecs/cng
* codecs/isac/fix
* codecs/isac/main

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

Review URL: https://webrtc-codereview.appspot.com/28699005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7432 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoWebRtcIsac_Decode et al.: Type encoded data as uint8[], not uint16[]
kwiberg@webrtc.org [Mon, 13 Oct 2014 11:23:24 +0000 (11:23 +0000)]
WebRtcIsac_Decode et al.: Type encoded data as uint8[], not uint16[]

This patch changes WebRtcIsac_Decode, WebRtcIsac_DecodeRcu, and
WebRtcIsacfix_Decode so that they read the encoded data from a uint8
array instead of a uint16 array.

BUG=909
R=aluebs@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7431 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoWebRtcIsac_UpdateBwEstimate et al.: Type byte streams as uint8, not uint16
kwiberg@webrtc.org [Mon, 13 Oct 2014 11:07:06 +0000 (11:07 +0000)]
WebRtcIsac_UpdateBwEstimate et al.: Type byte streams as uint8, not uint16

This patch changes the signature of WebRtcIsac_UpdateBwEstimate,
WebRtcIsacfix_UpdateBwEstimate, and WebRtcIsacfix_UpdateBwEstimate1 so
that they expect the encoded data to be uint8 arrays, not uint16,
which is more natural. The implementations of the functions are left
unchanged for now.

BUG=909
R=aluebs@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7430 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoSome WebRtcIsac_* and WebRtcIsacfix_* functions: type encoded stream as uint8[]
kwiberg@webrtc.org [Mon, 13 Oct 2014 10:53:42 +0000 (10:53 +0000)]
Some WebRtcIsac_* and WebRtcIsacfix_* functions: type encoded stream as uint8[]

The affected functions are

  WebRtcIsacfix_ReadFrameLen
  WebRtcIsacfix_GetNewBitStream
  WebRtcIsacfix_ReadBwIndex

and

  WebRtcIsac_ReadFrameLen
  WebRtcIsac_GetNewBitStream
  WebRtcIsac_ReadBwIndex
  WebRtcIsac_GetRedPayload

BUG=909
R=aluebs@webrtc.org, henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7429 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days ago(Auto)update libjingle 77414393-> 77554188
buildbot@webrtc.org [Mon, 13 Oct 2014 06:35:10 +0000 (06:35 +0000)]
(Auto)update libjingle 77414393-> 77554188

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7428 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoMerge the supporting to UYVY on Linux video capture in crbug/410202 to webrtc standalone.
braveyao@webrtc.org [Mon, 13 Oct 2014 02:13:00 +0000 (02:13 +0000)]
Merge the supporting to UYVY on Linux video capture in crbug/410202 to webrtc standalone.

BUG=3765
TEST=Manual
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7427 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoRelease _inputSendPin & _outputCapturePin before _captureFilter & _sinkFilter since...
braveyao@webrtc.org [Mon, 13 Oct 2014 02:11:55 +0000 (02:11 +0000)]
Release _inputSendPin & _outputCapturePin before _captureFilter & _sinkFilter since they should depend on the filters.
The previous steps work fine for all the webcam, but have problem on SplitCam driver as in the issue report.
Anyway it's always good to de-initial with the reversing order to initialization.

BUG=3845
TEST=Manual
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7426 4adac7df-926f-26a2-2b94-8c16560cd09d

13 days agoRe-enable ThreadCheckerDeathTest.MethodNotAllowedOnDifferentThreadInDebug (missed...
henrike@webrtc.org [Fri, 10 Oct 2014 21:41:55 +0000 (21:41 +0000)]
Re-enable ThreadCheckerDeathTest.MethodNotAllowedOnDifferentThreadInDebug (missed when enabling other base tests).

BUG=3836
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/24909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7425 4adac7df-926f-26a2-2b94-8c16560cd09d

13 days agoDisable SendsAndReceivesVP9 test for now.
marpan@webrtc.org [Fri, 10 Oct 2014 21:25:20 +0000 (21:25 +0000)]
Disable SendsAndReceivesVP9 test for now.

Fails on linux memcheck and DrMemory.
Will re-enable on next libvpx roll.

TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7424 4adac7df-926f-26a2-2b94-8c16560cd09d

13 days agoAdjust/increase rate control thresold for a vp9 test.
marpan@webrtc.org [Fri, 10 Oct 2014 17:55:57 +0000 (17:55 +0000)]
Adjust/increase rate control thresold for a vp9 test.

TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7423 4adac7df-926f-26a2-2b94-8c16560cd09d

13 days agoAdd VP9 codec to VCM and vie_auto_test.
marpan@webrtc.org [Fri, 10 Oct 2014 16:44:47 +0000 (16:44 +0000)]
Add VP9 codec to VCM and vie_auto_test.
Include VP9 tests in videoprocessor_integrationtests.
Include end-to-end send/receiveVP9 test.
Passes trybots.

R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7422 4adac7df-926f-26a2-2b94-8c16560cd09d

2 weeks agoMark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE.
xians@webrtc.org [Fri, 10 Oct 2014 09:42:53 +0000 (09:42 +0000)]
Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE.
This also marks all virtual overrides of other classes in the same files.

This will make a subsequent change I intend to do safer, where I'll change the
argument types of the base Transport functions, by breaking the compile if I
miss any overrides.

This also highlighted a number of unused functions. I've removed some of these.

TBR=mflodman@webrtc.org, pkasting@chromium.org
BUG=none
TEST=none

Review URL: https://webrtc-codereview.appspot.com/28709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7421 4adac7df-926f-26a2-2b94-8c16560cd09d

2 weeks agoCleanup scripts and suppressions for TSan v1
kjellander@webrtc.org [Fri, 10 Oct 2014 09:18:34 +0000 (09:18 +0000)]
Cleanup scripts and suppressions for TSan v1

Since we don't use it anymore on Linux and don't plan
to ever support it for Windows.

BUG=
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7420 4adac7df-926f-26a2-2b94-8c16560cd09d