Commit Graph

6830 Commits

Author SHA1 Message Date
henrik.lundin@webrtc.org
741711a861 Revert r7049/r7123, which added unnecessary "u"s to "return 0"s.
r7049 added some unnecessary casts ("return 0" -> "return static_cast<uint16_t>(0)").  r7123 converted these to "return 0u".  The original impetus for this was to eliminate type conversion warnings.  However, the 'u's are unnecessary; Visual Studio can return "0" from a function returning an unsigned value without producing a warning.  The real reason for the original warnings was that the code was returning -1 from a function returning an unsigned value, which does need a cast; the -1s were eliminated before the above two revisions landed.

Also reverse the order of some conditionals to prevent possible underflow.
While the underflow wouldn't have changed the behavior of the code, it's easier
to reason about the code when such underflow can't happen, and possibly safer
against future modifications as well.

BUG=3663
TEST=none
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7296 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-25 07:38:14 +00:00
pbos@webrtc.org
315669939a Fix typo from RtpPacketizerH264.
BUG=
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27609004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7295 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-25 07:31:42 +00:00
andresp@webrtc.org
37e1846d73 Revert "Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup." (rev 7293).
Breaks windows bot as it was already showing on the try jobs on the

BUG=crbug/414211
R=jiayl@webrtc.org,juberti@webrtc.org
TBR=jiayl@webrtc.org,juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7294 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-25 07:30:14 +00:00
jiayl@webrtc.org
fe1eafb71a Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup.
BUG=crbug/414211
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7293 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24 21:13:39 +00:00
andrew@webrtc.org
30be827e6a Enable render downmixing to mono in AudioProcessing.
In practice, we have been doing this since time immemorial, but have
relied on the user to do the downmixing (first voice engine then
Chromium). It's more logical for this burden to fall on AudioProcessing,
however, who can be expected to know that this is a reasonable approach
for AEC. Permitting two render channels results in running two AECs
serially.

Critically, in my recent change to have Chromium adopt the float
interface:
https://codereview.chromium.org/420603004
I removed the downmixing by Chromium, forgetting that we hadn't yet
enabled this feature in AudioProcessing. This corrects that oversight.

The change in paths hit by production users is very minor. As commented
it required adding downmixing to the int16_t path to satisfy
bit-exactness tests.

For reference, find the ApmTest.Process errors here:
https://paste.googleplex.com/6372007910309888

BUG=webrtc:3853
TESTED=listened to the files output from the Process test, and verified
that they sound as expected: higher echo while the AEC is adapting, but
afterwards very close.

R=aluebs@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7292 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24 20:06:23 +00:00
jiayl@webrtc.org
e1bba60792 Add missing DesktopConfigurationMonitor Unlock in webrtc::ScreenCapturerMac
BUG=3837
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7291 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24 17:23:46 +00:00
jiayl@webrtc.org
3987b6de50 Fix a problem in Thread::Send.
Previously if thread A->Send is called on thread B, B->ReceiveSends will be called, which enables an arbitrary thread to invoke calls on B while B is wait for A->Send to return. This caused mutliple problems like issue 3559, 3579.
The fix is to limit B->ReceiveSends to only process requests from A.
Also disallow the worker thread invoking other threads.

BUG=3559
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7290 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24 17:14:05 +00:00
aluebs@webrtc.org
a0ce9fa2a6 Call NS AnalyzeCaptureAudio before AEC
This attenuates the noise pumping generated from the NS adapting to the AEC comfort noise.

When there is echo present the AEC suppresses it and adds comfort noise. This is underestimated on purpose to avoid adding more than the original background noise. The NS has to be called after the AEC, because every non-linear processing before it can ruin its performance. Therefore the noise estimation can adapt to this comfort noise, making it less aggressive and generating noise pumping.

By putting the noise estimation analysis stage from the NS before the AEC, this effect can be avoided. This has been tested manually on recordings where noise pumping was present: Two long recordings done in our team by bjornv and kwiberg plus the most noisy (5) recordings in the QA set.

On the other hand, one risk of doing this is to not adapt to the comfort noise and therefore suppress too much. As verified in the tested files, this is not a problem in practice.

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

Review URL: https://webrtc-codereview.appspot.com/24679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7289 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24 14:18:03 +00:00
sprang@webrtc.org
70e2d11ea8 Reduce jitter delay for low fps streams.
Enabled by finch flag.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31389005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7288 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24 14:06:56 +00:00
aluebs@webrtc.org
275dac2c1d Moved the filter calculation from analyze to process in ns_core
It makes sense to have it there if the analyze and process methods are called in different stages.
Tested over the entire QA set for bit exactness.

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

Review URL: https://webrtc-codereview.appspot.com/29549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7287 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24 13:23:49 +00:00
bjornv@webrtc.org
634c926928 audioproc: Now also writes to output file in simulation mode
After changing to use wav as default file format no output was written in simulation mode.

BUG=3359
TESTED=locally
R=aluebs@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7286 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24 12:21:51 +00:00
kwiberg@webrtc.org
7ee24a7906 WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t
We have to fix both at once, since there's a macro that calls one of
them or the other.

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

Committed: https://code.google.com/p/webrtc/source/detail?r=7266

Review URL: https://webrtc-codereview.appspot.com/19229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7285 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24 10:31:02 +00:00
pbos@webrtc.org
d60d79a145 Thread annotation of rtc::CriticalSection.
Effectively re-lands r5516 which was reverted because talk/-only
checkouts existed. This now resides in webrtc/base/, so no talk/-only
checkouts should be possible.

This change also enables -Wthread-safety for talk/ and fixes a bug in
talk/media/webrtc/webrtcvideoengine2.cc where a guarded variable was
read without taking the corresponding lock.

R=andresp@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/27569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7284 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24 07:10:57 +00:00
pbos@webrtc.org
38344ed280 Move thread_annotations.h to webrtc/base/.
R=andresp@webrtc.org, mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/27579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7283 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24 06:05:00 +00:00
glaznev@webrtc.org
8166faeff3 Change Android video renderer to maintain video aspect
ratio when displaying camera or decoded video frames.

-

R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7282 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 23:58:52 +00:00
glaznev@webrtc.org
90668b1633 Switch HW video decoder to output byte buffers if video
renderer EGL context is not provided by app.

R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7281 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 21:42:15 +00:00
buildbot@webrtc.org
1b7dcc1647 (Auto)update libjingle 76169599-> 76176062
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7280 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 17:41:48 +00:00
johannkoenig@google.com
94ff92ceec Use VPX_IMG_FMT_*/VPX_PLANE_* defines
The compatibility layer has been removed upstream:
https://gerrit.chromium.org/gerrit/gitweb?p=webm%2Flibvpx.git;a=commit;h=9cdaa3d72eade9ad162ef8f78a93bd8f85c6de10

BUG=webrtc:3839
R=marpan@google.com, marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7279 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 17:31:47 +00:00
guoweis@webrtc.org
2c1bcea1bc Enable ipv6 by default for webrtc under a Finch experiment.
Reapply 23529005 after fixing the build break issue (Chromium:582133002)

Committed: https://code.google.com/p/webrtc/source/detail?r=7253

Review URL: https://webrtc-codereview.appspot.com/23529005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7278 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 16:23:02 +00:00
henrik.lundin@webrtc.org
3987f10c11 Revert "Remove DTMF status methods from Voice Engine" r7276
This change caused some trouble.

TBR=henrika@webrtc.org,pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7277 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 13:15:14 +00:00
henrik.lundin@webrtc.org
bf7b9e0081 Remove DTMF status methods from Voice Engine
These methods are not used.

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

Review URL: https://webrtc-codereview.appspot.com/24689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7276 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 12:54:04 +00:00
kjellander@webrtc.org
e34a2e7475 Revert "Set minimum SDK level to 10.7 for Mac and iOS" (r7175)
Reverting this since it didn't fix the build failures.
We ended up passing mac_sdk=10.9 in GYP_DEFINES on the bots
to to make the build pass again
(https://codereview.chromium.org/573673002).

BUG=3120
R=mcasas@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7275 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 12:43:14 +00:00
pbos@webrtc.org
faf2410a32 gn: Hide modules/video_capture:video_capture_internal_impl behind an arg
R=andresp@webrtc.org, brettw@chromium.org, kjellander@webrtc.org, pbos@webrtc.org, brettw

Review URL: https://webrtc-codereview.appspot.com/30479004

Patch from Cem Kocagil <ckocagil@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7274 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 12:37:06 +00:00
henrik.lundin@webrtc.org
0e6e4d2ff2 Reland "Converting five tests to use new AudioCoding interface" (r7258)
This CL reverts r7264. The problem was that iSAC-SWB and iSAC-FB are
not supported on android. These are now disabled.

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7273 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 12:05:34 +00:00
andresp@webrtc.org
4f6f22f0c6 Reland (rev 7259) "Convert AcmReceiverTest to new AudioCoding interface"
Was reverted by mistake in 7260. Actual culprit was 7258.

BUG=3520
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7272 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 11:37:57 +00:00
bjornv@webrtc.org
ea29787df0 audio_processing/agc: Solved building with AGC_DEBUG + few style changes
webrtc did not build if AGC_DEBUG was turned on. This CL fixes that. Has no impact on performance since it is development/debug code.

* Name change to WEBRT_AGC_DEBUG_DUMP
* Added build flag agc_debug_dump to .gypi
* Added missing "%d" in printf at two places
* Some line length related style changes

Tested audioproc and modules_unittests with GYP_DEFINES=agc_debug_dump=1 webrtc/build/gyp_webrtc

BUG=N/A
TESTED=locally and trybots
R=aluebs@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7271 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 11:21:39 +00:00
pbos@webrtc.org
0a2087a711 Skeleton for registering external encoders/decoders.
R=pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/31429005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7270 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 09:40:22 +00:00
tkchin@webrtc.org
c569a49a3d Unit tests for SSLAdapter
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17309004

Patch from Manish Jethani <manish.jethani@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7269 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 05:56:44 +00:00
bjornv@webrtc.org
dc0b37dcb1 modules_unittests: Turned on ApmTest.Process test for Android
The reason why ApmTest.Process breaks on Android is that two metrics over counts. I decided to add an offset and a different slack to the EXPECT_NEAR() calls that are affected. I think this is a reasonable approach since we have no more than two failing metrics. If any feature change that will make another metric fail, we should go back to the desk and find another way of solving this.

BUG=114
TESTED=locally on Nexus 7 and trybots
R=aluebs@webrtc.org, andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7268 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 05:03:44 +00:00
andrew@webrtc.org
a3c4d4dd2c Revert 7266 "WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type en..."
This was causing apparently legitimate failures on the following bots:
http://chromegw/i/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/2599
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28KK%20Nexus5%29%28dbg%29/builds/2023
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28JB%20Nexus7.2%29%28dbg%29/builds/1825
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28KK%20Nexus5%29/builds/2013
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28JB%20Nexus7.2%29/builds/1795

> WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t
> 
> We have to fix both at once, since there's a macro that calls one of
> them or the other.
> 
> BUG=909
> R=andrew@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, minyue@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/19229004

TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7267 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 01:32:57 +00:00
kwiberg@webrtc.org
8c5740b485 WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t
We have to fix both at once, since there's a macro that calls one of
them or the other.

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

Review URL: https://webrtc-codereview.appspot.com/19229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7266 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-22 23:04:14 +00:00
pbos@webrtc.org
83f95ba9a6 Remove engine-level SetOptions.
Already removed in WebRtcVideoEngine.

R=andresp@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/29549005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7265 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-22 16:07:18 +00:00
andresp@webrtc.org
99e404c84a Revert "Converting five tests to use new AudioCoding interface" (rev 7258).
This time reverts the Cl that actually broke the tests. Got the wrong rev before. :/

BUG=3520
TESTED=Locally with CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py gtest -s modules_unittests --gtest_filter=AcmReceiverBitExactness.8kHzOutput --verbose --isolate-file-path=webrtc/modules/modules_unittests.isolate
TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7264 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-22 15:49:56 +00:00
houssainy@google.com
35850ff71f Adding test file path as argument of the rtcBot run command's arguments.
The new command to run rtcBot is:-
node test.js <bot_type> <test_file_path>

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7263 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-22 15:24:56 +00:00
henrik.lundin@webrtc.org
64a2f10f4b Remove Get/SetNetEQPlayoutMode APIs
These are not used anymore.

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

Review URL: https://webrtc-codereview.appspot.com/26569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7262 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-22 14:30:10 +00:00
houssainy@google.com
07ca949346 Adding webrtc_video_streaming test
This test is streaming video and audio between two bots using webrtc js api.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7261 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-22 13:52:39 +00:00
andresp@webrtc.org
c570761288 Revert "Convert AcmReceiverTest to new AudioCoding interface" (rev 7258).
Breaks android modules_unittests tests by crashing on AcmReceiverBitExactness.8kHzOutput
Was already visible on "git cl try" before submitting on https://webrtc-codereview.appspot.com/23719004/#

BUG=3520
R=kwiberg@webrtc.org, henrik.lundin@webrtc.org
TBR=kwiberg@webrtc.org, henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7260 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-22 13:18:34 +00:00
henrik.lundin@webrtc.org
cfe073539c Convert AcmReceiverTest to new AudioCoding interface
In order to maintain test coverage for the old API (AudioCodingModule)
during the transition period, the old test was copied to
AcmReceiverTestOldApi.

Modified and extended AudioCoding and the implementation to make the
test compile and run.

Created a converter method from new to old config struct

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7259 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-22 12:10:44 +00:00
henrik.lundin@webrtc.org
eb1de5cb72 Converting five tests to use new AudioCoding interface
The converted tests are:
AcmIsacMtTest
AcmReceiverBitExactness
AcmSenderBitExactness
AudioCodingModuleMtTest
AudioCodingModuleTest

In order to maintain test coverage for the old API (AudioCodingModule)
during the transition period, the old tests were copied and given the
suffix OldApi:
AcmIsacMtTestOldApi
AcmReceiverBitExactnessOldApi
AcmSenderBitExactnessOldApi
AudioCodingModuleMtTestOldApi
AudioCodingModuleTestOldApi

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7258 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-22 12:07:12 +00:00
aluebs@webrtc.org
bdfdc96b22 Clang-format ns_core
BUG=webrtc:3811
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7257 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-22 10:59:46 +00:00
pbos@webrtc.org
759982d357 Set number of temporal layers for VideoSendStream.
Introduces a mapping between EncoderConfig and VideoCodec. More
specifically it also removes an assert that there should be no set
temporal layers in the new API, which is wrong and was temporary.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/25619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7256 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-22 09:32:46 +00:00
henrik.lundin@webrtc.org
612171527e Ensure that NetEq recovers after a large timestamp jump
Before this change it could happen that a large jump in timestamp (a
jump not correlated to wall-clock change) caused the audio to go silent
without recovering. The reason was that all incoming packets after the
jump were considered too old compared to the last decoded packet, and
were deleted. With CL changes two things:

1. If the only available packet in the buffer is an old packet, NetEq
will do Expand instead of immediate reset. This is to avoid that one
late packet triggers a reset.

2. Old packets are discarded only when the decision to decode a packet
has been taken. This is to allow the buffer to grow and eventually
flush if no decodable packet has been found for some time.

This CL also includes a new unit test for this situation.

BUG=3785
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7255 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-22 08:30:07 +00:00
henrike@webrtc.org
88772874da Disabled several rtc_unittests so the tests can be turned on in the waterfall
BUG=3836
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7254 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-22 07:30:48 +00:00
guoweis@webrtc.org
97ed39344a Reapply 23529005 after fixing the build break issue (Chromium:582133002)
Review URL: https://webrtc-codereview.appspot.com/23529005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7253 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-19 21:06:12 +00:00
buildbot@webrtc.org
ed5ca1f122 (Auto)update libjingle 75925673-> 75926712
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7252 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-19 20:30:44 +00:00
buildbot@webrtc.org
c98f217c65 (Auto)update libjingle 75924589-> 75925673
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7251 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-19 20:18:10 +00:00
buildbot@webrtc.org
0c9fe72b21 (Auto)update libjingle 75922684-> 75924589
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7250 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-19 20:05:02 +00:00
glaznev@webrtc.org
ebf2757339 Fix HW video decoder crash on some Android KK devices.
Remove direct access to decoder Java output buffer memory
when HW decoder is configured to decode to surface.

-

R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30459005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7249 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-19 19:36:13 +00:00
thorcarpenter@google.com
c1eebfa107 Fix the libjingle_media_unittest failure in Windows build by modifying libjingle_tests.gyp and sctpdataengine_unittests.cc instead of ssladapter.cc.
R=harryjin@google.com, pthatcher@webrtc.org, tpsiaki@google.com

Review URL: https://webrtc-codereview.appspot.com/22699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7245 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-19 17:54:00 +00:00
glaznev@webrtc.org
e65812427d Fixing compilation failure in peerconnection_jni.cc with WEBRTC_CHROMIUM_BUILD.
Symbol LogcatTraceContext not defined.
Submitting on behalf of serya@.
Dup of https://webrtc-codereview.appspot.com/29529004/

TEST=Build target libjingle_peerconnection_javalib with applied CL https://codereview.chromium.org/551793003/
BUG=https://crbug.com/383418
R=serya@chromium.org

Review URL: https://webrtc-codereview.appspot.com/28529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7244 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-19 16:53:46 +00:00