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Author SHA1 Message Date
buildbot@webrtc.org
81ddc78536 (Auto)update libjingle 77701902-> 77709729 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7450 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 22:39:24 +00:00
buildbot@webrtc.org
1ecbe45c7e (Auto)update libjingle 77689511-> 77696841 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7449 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 20:29:28 +00:00
pbos@webrtc.org
43336b6b9f Remove unused (no-op) VideoOptions. 
Removing VideoOptions: adapt_input_to_encoder, adapt_view_switch,
video_one_layer_screencast and video_high_bitrate.

R=pthatcher@webrtc.org
BUG=

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7448 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 19:12:06 +00:00
henrike@webrtc.org
a4351a045d 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
 2014-10-14 17:07:41 +00:00
pbos@webrtc.org
a73a678e25 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
 2014-10-14 11:52:10 +00:00
stefan@webrtc.org
eb24b04f16 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
 2014-10-14 11:40:13 +00:00
henrik.lundin@webrtc.org
81a78930ee 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
 2014-10-14 10:49:58 +00:00
stefan@webrtc.org
c216b9aeaf 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
 2014-10-14 10:38:49 +00:00
kwiberg@webrtc.org
a57678a70e 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
 2014-10-14 09:40:04 +00:00
pbos@webrtc.org
7fe1e03dd6 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
 2014-10-14 04:25:33 +00:00
buildbot@webrtc.org
f68cc0b0c3 (Auto)update libjingle 77554188-> 77629208 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7439 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 01:17:42 +00:00
marpan@webrtc.org
82e6fa533c 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
 2014-10-14 00:34:19 +00:00
aluebs@webrtc.org
b6af4283ca 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
 2014-10-13 20:48:05 +00:00
henrike@webrtc.org
1e6a5dd14e 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
 2014-10-13 18:27:11 +00:00
marpan@webrtc.org
8bee130fa0 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
 2014-10-13 17:10:40 +00:00
bjornv@webrtc.org
bc1a4578e0 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
 2014-10-13 14:00:43 +00:00
kwiberg@webrtc.org
a3722b643d 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
 2014-10-13 13:29:04 +00:00
bjornv@webrtc.org
d4fe824862 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
 2014-10-13 13:01:13 +00:00
kwiberg@webrtc.org
396a5e0001 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
 2014-10-13 11:23:24 +00:00
kwiberg@webrtc.org
3f7f899a15 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
 2014-10-13 11:07:06 +00:00
kwiberg@webrtc.org
1172988c79 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
 2014-10-13 10:53:42 +00:00
buildbot@webrtc.org
3c16d8bd1c (Auto)update libjingle 77414393-> 77554188 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7428 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-13 06:35:10 +00:00
braveyao@webrtc.org
c502df54f8 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
 2014-10-13 02:13:00 +00:00
braveyao@webrtc.org
651c05e4fc 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
 2014-10-13 02:11:55 +00:00
henrike@webrtc.org
7f7b0a1cdd 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
 2014-10-10 21:41:55 +00:00
marpan@webrtc.org
4ddbbed16e 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
 2014-10-10 21:25:20 +00:00
marpan@webrtc.org
c87b74717b 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
 2014-10-10 17:55:57 +00:00
marpan@webrtc.org
573c78e31c 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
 2014-10-10 16:44:47 +00:00
xians@webrtc.org
3cefbc99f4 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
 2014-10-10 09:42:53 +00:00
kjellander@webrtc.org
afede835ee 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
 2014-10-10 09:18:34 +00:00
pbos@webrtc.org
fae6bc4106 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
 2014-10-10 08:45:03 +00:00
xians@webrtc.org
e46bc77e94 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
 2014-10-10 08:36:56 +00:00
kjellander@webrtc.org
c5593ef1aa 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
 2014-10-10 07:16:05 +00:00
henrike@webrtc.org
c732a3e511 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
 2014-10-09 22:08:15 +00:00
henrike@webrtc.org
4a73519690 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
 2014-10-09 20:27:13 +00:00
glaznev@webrtc.org
dae40dcde9 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
 2014-10-09 17:53:09 +00:00
henrike@webrtc.org
e30dab77df 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
 2014-10-09 15:41:40 +00:00
henrike@webrtc.org
17f8ddd6c4 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
 2014-10-09 15:40:18 +00:00
henrik.lundin@webrtc.org
8768f161cd 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
 2014-10-09 12:58:45 +00:00
pbos@webrtc.org
3ff788cf73 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
 2014-10-09 12:47:15 +00:00
kwiberg@webrtc.org
4bd2db9a55 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
 2014-10-09 11:21:10 +00:00
kjellander@webrtc.org
1bada48401 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
 2014-10-09 10:53:02 +00:00
minyue@webrtc.org
2c0cdbce22 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
 2014-10-09 10:52:43 +00:00
minyue@webrtc.org
c803907d87 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
 2014-10-09 10:49:54 +00:00
kjellander@webrtc.org
0b0ac8236b 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
 2014-10-09 09:11:27 +00:00
bjornv@webrtc.org
3ea35fdb1b 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
 2014-10-09 08:47:02 +00:00
pbos@webrtc.org
127ca3f8e5 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
 2014-10-09 07:52:03 +00:00
andrew@webrtc.org
0001adcfef 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
 2014-10-09 04:13:02 +00:00
glaznev@webrtc.org
95bacfed08 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
 2014-10-09 00:00:11 +00:00
buildbot@webrtc.org
97abeee282 (Auto)update libjingle 77263371-> 77296420 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7400 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-08 22:24:30 +00:00