external/webrtc.git
2 hours agoImplement AudioEncoderPcmU/A classes and convert AudioDecoder tests master
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

7 hours 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

8 hours 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

19 hours 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

24 hours 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

28 hours 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

29 hours 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

30 hours 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

30 hours 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

3 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

3 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

3 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

3 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

3 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

3 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

3 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

4 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

4 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

5 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

5 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

5 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

5 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

5 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

6 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

6 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

6 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

6 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

6 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

6 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

6 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

6 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

6 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

6 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

6 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

6 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

7 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

7 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

7 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

7 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

7 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

7 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

7 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

7 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

7 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

7 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

7 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

8 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

8 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

8 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

8 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

8 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

8 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

8 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

8 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

8 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

10 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

10 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

10 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

10 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

11 days 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

11 days 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

11 days agoRemove talk_base from suppressions.
pbos@webrtc.org [Fri, 10 Oct 2014 08:45:03 +0000 (08:45 +0000)]
Remove talk_base from suppressions.

This namespace doesn't exist anymore, so remove all suppressions that
include it in the call stack.

R=kjellander@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/31639005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7419 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoReland 28629004: adding new AEC dump start interface for chrome.
xians@webrtc.org [Fri, 10 Oct 2014 08:36:56 +0000 (08:36 +0000)]
Reland 28629004: adding new AEC dump start interface for chrome.

This is required because we are not allow to pass CRT objects across dll boundaries, that says, when we pass a file descriptor from chrome dll to libpeerconnection dll, the file descriptor will become invalid immediate, more information can be found here:
http://msdn.microsoft.com/en-us/library/ms235460.aspx

R=andresp@webrtc.org, andrew@webrtc.org, bjornv@webrtc.org, henrike@webrtc.org, henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7418 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoWorkaround deps2git issue with inline Python in DEPS.
kjellander@webrtc.org [Fri, 10 Oct 2014 07:16:05 +0000 (07:16 +0000)]
Workaround deps2git issue with inline Python in DEPS.

When running
https://code.google.com/p/chromium/codesearch#chromium/tools/deps2git/deps2git.py
on our DEPS file, an error is caused by the formatting pretty printing
of the converted DEPS -> .DEPS.git output.
Since this needs to work in order to switch our bots to bot_update
(uses Git) and the fact that changing deps2git.py is high risk, it's
better to work around this problem by altering the Python inline code.
The fact that deps2git will go away when the remaining projects
have switched to Git also motivates not taking the risk of changing
deps2git for this case only.

BUG=3534
TESTED=Ran gclient runhooks and verified the script executed
when there was a trunk/check_root_dir.py file.

R=phoglund@chromium.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7417 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoRe-enable allmost all base tests.
henrike@webrtc.org [Thu, 9 Oct 2014 22:08:15 +0000 (22:08 +0000)]
Re-enable allmost all base tests.

BUG=3836
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/22989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7416 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoRe-enables a bunch of base unittests part II.
henrike@webrtc.org [Thu, 9 Oct 2014 20:27:13 +0000 (20:27 +0000)]
Re-enables a bunch of base unittests part II.

BUG=3836
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/30709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7415 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoChange setting VP8 codec specific info values by HW VP8 encoder
glaznev@webrtc.org [Thu, 9 Oct 2014 17:53:09 +0000 (17:53 +0000)]
Change setting VP8 codec specific info values by HW VP8 encoder
to follow SW implementation.

This fixes video freezing observed when connecting Android AppRtcDemo
on devices with hW encoder support with Chrome apprtc.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7414 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agobase/thread_unittest: wrap test was setting current thread to NULL.
henrike@webrtc.org [Thu, 9 Oct 2014 15:41:40 +0000 (15:41 +0000)]
base/thread_unittest: wrap test was setting current thread to NULL.

This broke unittests following ThreadTest.Wrap

BUG=3836
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7413 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoMake pbos and kjellander only owners of tsan2 suppressions.
henrike@webrtc.org [Thu, 9 Oct 2014 15:40:18 +0000 (15:40 +0000)]
Make pbos and kjellander only owners of tsan2 suppressions.

R=pbos@webrtc.org
TBR=kjellander@webrtc.org
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/29709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7412 4adac7df-926f-26a2-2b94-8c16560cd09d

12 days agoFix comments in common_types.h
henrik.lundin@webrtc.org [Thu, 9 Oct 2014 12:58:45 +0000 (12:58 +0000)]
Fix comments in common_types.h

Two of the metrics in NetworkStatistics were desribed as being in
percent, while they are in fact fractions between 0 and 1, scaled
to Q14 domain.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7411 4adac7df-926f-26a2-2b94-8c16560cd09d

12 days agoIncrease timeout for AsyncWriteTest.TestWrite.
pbos@webrtc.org [Thu, 9 Oct 2014 12:47:15 +0000 (12:47 +0000)]
Increase timeout for AsyncWriteTest.TestWrite.

Having a 10ms timeout for something meant to run on DrMemory is insane.

TBR=henrike@webrtc.org
BUG=3490

Review URL: https://webrtc-codereview.appspot.com/23959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7410 4adac7df-926f-26a2-2b94-8c16560cd09d

12 days agoOpus wrapper: Use const for inputs and uint8[] for byte streams
kwiberg@webrtc.org [Thu, 9 Oct 2014 11:21:10 +0000 (11:21 +0000)]
Opus wrapper: Use const for inputs and uint8[] for byte streams

About half of the functions already followed the desired pattern; this
patch fixes the other half.

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

Review URL: https://webrtc-codereview.appspot.com/26719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7409 4adac7df-926f-26a2-2b94-8c16560cd09d

12 days agoMake DEPS find check_root_dir.py in legacy checkouts.
kjellander@webrtc.org [Thu, 9 Oct 2014 10:53:02 +0000 (10:53 +0000)]
Make DEPS find check_root_dir.py in legacy checkouts.

In r7405 the DEPS hook wasn't properly handling the case
when the trunk dir is not yet renamed. This makes the script
only be called if it exists in the old not-yet-renamed trunk dir.

BUG=3534
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7408 4adac7df-926f-26a2-2b94-8c16560cd09d

12 days agoEstimating NTP time with a given RTT.
minyue@webrtc.org [Thu, 9 Oct 2014 10:52:43 +0000 (10:52 +0000)]
Estimating NTP time with a given RTT.

RemoteNtpTimeEstimator needed user to give a remote SSRC and it intended to call RtpRtcp module to obtain RTT, to be able to calculate Ntp time.

When RTT cannot be directly obtained from the RtpRtcp module with the specified SSRC, RemoteNtpTimeEstimator would fail.

This change allows RemoteNtpTimeEstimator to calculate NTP with an external RTT estimate.

An immediate benefit is that capture_start_ntp_time_ms_ can be obtained in a Google hangout call.

BUG=

TEST=chromium + hangout call
R=stefan@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7407 4adac7df-926f-26a2-2b94-8c16560cd09d

12 days agoRemoving useless packets when inserting them (NetEq)
minyue@webrtc.org [Thu, 9 Oct 2014 10:49:54 +0000 (10:49 +0000)]
Removing useless packets when inserting them (NetEq)

This is to save the buffer.

Some old code may become unnecessary, and will be removed in a separate CL.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7406 4adac7df-926f-26a2-2b94-8c16560cd09d

12 days agoRemove root_dir variable from DEPS + enforce rename.
kjellander@webrtc.org [Thu, 9 Oct 2014 09:11:27 +0000 (09:11 +0000)]
Remove root_dir variable from DEPS + enforce rename.

Update DEPS to no longer have the root_dir variable.
Add a script that detects if the user have a solution named
'trunk' and explains what needs to be done to change it to 'src'.

The reason for this change is that bot_update doesn't support
custom_vars in solutions and Chrome infra is trying to get
rid of them entirely in the future.

The bots are already using a solution named 'src' so they
won't run into this error during runhooks.

BUG=3534
TESTED=Ran the script with a .gclient containing a solution
named 'trunk' and one named 'src'. Also tested the presence
of the custom_vars dict and not.

R=andrew@webrtc.org, hinoka@chromium.org

Review URL: https://webrtc-codereview.appspot.com/30619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7405 4adac7df-926f-26a2-2b94-8c16560cd09d

12 days agocommon_audio: Removed macro WEBRTC_SPL_LSHIFT_W16
bjornv@webrtc.org [Thu, 9 Oct 2014 08:47:02 +0000 (08:47 +0000)]
common_audio: Removed macro WEBRTC_SPL_LSHIFT_W16

The macro was a trivial << operation and where used has been replaced by <<. Affected components are
* ilbc
* isacfix

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

Review URL: https://webrtc-codereview.appspot.com/22919005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7404 4adac7df-926f-26a2-2b94-8c16560cd09d

12 days agoDisable TestDTLSConnectWithSmallMtu on all platforms.
pbos@webrtc.org [Thu, 9 Oct 2014 07:52:03 +0000 (07:52 +0000)]
Disable TestDTLSConnectWithSmallMtu on all platforms.

Other bots elsewhere are breaking on this test, my money is on that this
might be due to different SSL versions being used on the different bots.
This test fails on at least a couple of bots that has use_openssl=1.

R=kjellander@webrtc.org
TBR=henrike@webrtc.org
BUG=3910

Review URL: https://webrtc-codereview.appspot.com/25839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7403 4adac7df-926f-26a2-2b94-8c16560cd09d

12 days agoUse openmax_dl on all ARM (v7 or higher) platforms.
andrew@webrtc.org [Thu, 9 Oct 2014 04:13:02 +0000 (04:13 +0000)]
Use openmax_dl on all ARM (v7 or higher) platforms.

openmax_dl now works on non-Android ARM, but it still requires
arm_version >= 7, and doesn't work on iOS at all.

TEST=Chromium build for a ChromeOS ARM device passes.
BUG=chromium:415393
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7402 4adac7df-926f-26a2-2b94-8c16560cd09d

12 days agoRemove bad waiting code from video decoder release function.
glaznev@webrtc.org [Thu, 9 Oct 2014 00:00:11 +0000 (00:00 +0000)]
Remove bad waiting code from video decoder release function.

Instead keep surface texture object alive while video codec
is re-initialized with a different resolution.

BUG=
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7401 4adac7df-926f-26a2-2b94-8c16560cd09d

12 days ago(Auto)update libjingle 77263371-> 77296420
buildbot@webrtc.org [Wed, 8 Oct 2014 22:24:30 +0000 (22:24 +0000)]
(Auto)update libjingle 77263371-> 77296420

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7400 4adac7df-926f-26a2-2b94-8c16560cd09d

12 days agoRe-enables a bunch of base unittests.
henrike@webrtc.org [Wed, 8 Oct 2014 22:17:02 +0000 (22:17 +0000)]
Re-enables a bunch of base unittests.

BUG=3836
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7399 4adac7df-926f-26a2-2b94-8c16560cd09d

12 days agoRoll chromium_revision fc668e2..2d714fa (298195:298667)
andrew@webrtc.org [Wed, 8 Oct 2014 19:16:10 +0000 (19:16 +0000)]
Roll chromium_revision fc668e2..2d714fa (298195:298667)

Picks up openmax_dl fixes for non-Android ARM.

Summary of changes (git diff fc668e2..2d714fa DEPS):
* third_party/boringssl c7dd5f3..51fcd87
* third_party/openmax_dl/dl/src 79e64bc..0164270
* third_party/usrsctp/usrsctplib d5685d4..dfd687b
* tools/swarming_client 33d442a..c28b74f

TBR=kjellander
BUG=chromium:415393,webrtc:3906

Review URL: https://webrtc-codereview.appspot.com/23929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7398 4adac7df-926f-26a2-2b94-8c16560cd09d

12 days agoAdd a variable for deciding when to use openmax_dl.
andrew@webrtc.org [Wed, 8 Oct 2014 18:01:27 +0000 (18:01 +0000)]
Add a variable for deciding when to use openmax_dl.

Modifies the previous condition to additionally not use openmax_dl on
iOS. Remove the All target's direct dependency on it, as it is now
pulled in by the targets that need it.

Add gn support since an openmax_dl gn target is available.

BUG=chromium:415393, webrtc:3906
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7397 4adac7df-926f-26a2-2b94-8c16560cd09d

12 days agoaudio_coding: Replaced macro WEBRTC_SPL_RSHIFT_W16 with >>
bjornv@webrtc.org [Wed, 8 Oct 2014 15:36:30 +0000 (15:36 +0000)]
audio_coding: Replaced macro WEBRTC_SPL_RSHIFT_W16 with >>

Replaced trivial shift macro with >>. The actual implementation of the macro is simply >>.

Affected codecs:
* ilbc
* isac/fix

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

Review URL: https://webrtc-codereview.appspot.com/23889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7396 4adac7df-926f-26a2-2b94-8c16560cd09d

13 days agoProtect send_/recv_streams_ in WebRtcVideoEngine2.
pbos@webrtc.org [Wed, 8 Oct 2014 14:48:08 +0000 (14:48 +0000)]
Protect send_/recv_streams_ in WebRtcVideoEngine2.

Important as OnLoadUpdate() won't be called on the worker thread and the
list of streams can't be concurrently modified while delivering this
callback to all send streams.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/22959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7395 4adac7df-926f-26a2-2b94-8c16560cd09d

13 days agoCHECK/DCHECK: Explicitly state whether the condition can have side effects
kwiberg@webrtc.org [Wed, 8 Oct 2014 12:19:56 +0000 (12:19 +0000)]
CHECK/DCHECK: Explicitly state whether the condition can have side effects

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7394 4adac7df-926f-26a2-2b94-8c16560cd09d

13 days agoChange name of a NetEq internal member variable
henrik.lundin@webrtc.org [Wed, 8 Oct 2014 12:10:53 +0000 (12:10 +0000)]
Change name of a NetEq internal member variable

In the StatisticsCalculator class, the member last_report_timestamp_
was unfortunately named. It's now been changed to
timestamps_since_last_report_, which describes it more accurately.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7393 4adac7df-926f-26a2-2b94-8c16560cd09d

13 days agoMake the media content send only if offerToReceive is false while local streams exist.
jiayl@webrtc.org [Tue, 7 Oct 2014 21:32:43 +0000 (21:32 +0000)]
Make the media content send only if offerToReceive is false while local streams exist.
We previously do not add the media content if offerToReceive is false.

BUG=3833
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7390 4adac7df-926f-26a2-2b94-8c16560cd09d

13 days agoInitialize sctp_paddrparams in OpenSctpSocket().
pbos@webrtc.org [Tue, 7 Oct 2014 19:23:43 +0000 (19:23 +0000)]
Initialize sctp_paddrparams in OpenSctpSocket().

Addresses 'use-of-uninitialized-value' detected with MemorySanitizer.
params.spp_address.sa_family was used without being initialized before
when calling usrsctp_setsockopt with SCTP_PEER_ADDR_PARAMS.

R=jiayl@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/23909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7389 4adac7df-926f-26a2-2b94-8c16560cd09d

13 days agoExplicitly unpoison FDs for MSan.
pbos@webrtc.org [Tue, 7 Oct 2014 17:56:53 +0000 (17:56 +0000)]
Explicitly unpoison FDs for MSan.

MSan doesn't handle inline assembly that's used by FD_ZERO causing a
false positive.

R=earthdok@chromium.org, henrike@webrtc.org
BUG=chromium:344505

Review URL: https://webrtc-codereview.appspot.com/25799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7388 4adac7df-926f-26a2-2b94-8c16560cd09d

13 days agoTemporary fix to allow Invoke() calls for VP8 HW encoder and decoder.
glaznev@webrtc.org [Tue, 7 Oct 2014 17:11:36 +0000 (17:11 +0000)]
Temporary fix to allow Invoke() calls for VP8 HW encoder and decoder.

BUG=
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7387 4adac7df-926f-26a2-2b94-8c16560cd09d

2 weeks agoRemove potential deadlock in WebRtcVideoEngine2.
pbos@webrtc.org [Tue, 7 Oct 2014 14:27:27 +0000 (14:27 +0000)]
Remove potential deadlock in WebRtcVideoEngine2.

Fixes lock-order inversions between capturer's SignalVideoFrame and
WebRtcVideoSendStream. Additionally also removes all deadlock
suppressions for WebRtcVideoEngine2.

R=stefan@webrtc.org
TBR=kjellander@webrtc.org
BUG=1788,2999

Review URL: https://webrtc-codereview.appspot.com/26729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7386 4adac7df-926f-26a2-2b94-8c16560cd09d

2 weeks agoRoll chromium_revision c264a05..fc668e2 (297113:298195)
kjellander@webrtc.org [Tue, 7 Oct 2014 12:49:34 +0000 (12:49 +0000)]
Roll chromium_revision c264a05..fc668e2 (297113:298195)

Mainly to pick up https://codereview.chromium.org/619723006/
to fix our MSan bot.

Summary of changes (git diff c264a05..fc668e2 DEPS):
* third_party/boringssl 01fe820..c7dd5f30
* third_party/usrsctp/usrsctplib 8975bd5..d5685d4
* tools/swarming_client 79940aee..33d442a

Clang updated 216630:217949 (git diff c264a05..fc668e2 tools/clang/scripts/update.sh)
This caused TSan v2 to hit an assert in libjingle_peerconnection_unittest
and libjingle_media_unittest, which is why the dlclose call
had to be disabled for now (webrtc:3895).

BUG=3895
R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7385 4adac7df-926f-26a2-2b94-8c16560cd09d

2 weeks agoRevert "Only configure the SSL library in one place."
pbos@webrtc.org [Tue, 7 Oct 2014 11:43:03 +0000 (11:43 +0000)]
Revert "Only configure the SSL library in one place."

This reverts commit r7378, which broke Windows compile targets
elsewhere.

R=kjellander@webrtc.org
TBR=henrike@webrtc.org
BUG=chromium:413497

Review URL: https://webrtc-codereview.appspot.com/28679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7384 4adac7df-926f-26a2-2b94-8c16560cd09d

2 weeks agoIsolate: Remove use of --ignore_broken_items
kjellander@webrtc.org [Tue, 7 Oct 2014 09:17:35 +0000 (09:17 +0000)]
Isolate: Remove use of --ignore_broken_items

BUG=chromium:395700
R=jam@chromium.org

Review URL: https://webrtc-codereview.appspot.com/30659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7383 4adac7df-926f-26a2-2b94-8c16560cd09d

2 weeks agoFix neteq_rtpplay so that empty SSRC is valid
henrik.lundin@webrtc.org [Tue, 7 Oct 2014 07:18:36 +0000 (07:18 +0000)]
Fix neteq_rtpplay so that empty SSRC is valid

In r7380, the command line flag --ssrc was added to neteq_rtpplay.
However, it was not possible to omit that flag, since the validation
did not accept an empty string. This CL fixes that.

TBR=kwiberg@webrtc.org
BUG=2692

Review URL: https://webrtc-codereview.appspot.com/24869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7382 4adac7df-926f-26a2-2b94-8c16560cd09d

2 weeks agoSet NetEq playout mode through the Config struct
henrik.lundin@webrtc.org [Tue, 7 Oct 2014 06:37:39 +0000 (06:37 +0000)]
Set NetEq playout mode through the Config struct

This change opens up the possibility to set the playout mode when
creating the NetEq object. The old methods SetPlayoutMode and
PlayoutMode are still available, but are deprecated.

BUG=3520
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7381 4adac7df-926f-26a2-2b94-8c16560cd09d

2 weeks agoAdd an SSRC filter to neteq_rtpplay
henrik.lundin@webrtc.org [Tue, 7 Oct 2014 05:30:04 +0000 (05:30 +0000)]
Add an SSRC filter to neteq_rtpplay

This allows the user to set the desired SSRC if the input file
contains multiple streams.

BUG=2692
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7380 4adac7df-926f-26a2-2b94-8c16560cd09d

2 weeks agoPrevent reading outside iSAC bitstream, if the stream is corrupted.
turaj@webrtc.org [Tue, 7 Oct 2014 00:21:02 +0000 (00:21 +0000)]
Prevent reading outside iSAC bitstream, if the stream is corrupted.

BUG=chrome_373312(#24)
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7379 4adac7df-926f-26a2-2b94-8c16560cd09d