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Author SHA1 Message Date
henrik.lundin@webrtc.org
5ecdef11cc Do not use recursive calling in NetEq test tools 
This CL removes recursive calling in:
- NETEQTEST_DummyRTPpacket::readFromFile,
- NETEQTEST_RTPpacket::readFromFile.

The files currently exist for both NetEq3 and NetEq4, and all are
changed with this CL.

R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5200 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-03 08:26:49 +00:00
fischman@webrtc.org
e0034557a7 RTCPeerConnection(objc): avoid leaking ICE candidate on addition. 
BUG=2670
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5199 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-02 18:49:54 +00:00
tina.legrand@webrtc.org
8418e9696b Fixing NetEq tests for new Opus version 
The new version of Opus doesn't generate the same number of bytes encoding the test vectors in audio_decoder_unittest. Therefore the test was updated not to check the length of the encoded packet, to prepare for the coming roll of Opus. Same change was applied to iSAC, which can also generate different number of bytes on different platforms.

BUG=1459
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5195 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-29 09:30:43 +00:00
braveyao@webrtc.org
54e8bfafba Apprtc demo: add DSCP support. 
BUG=2669
TEST=Manual Test
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5194 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-29 02:38:20 +00:00
phoglund@webrtc.org
03c7a35ac0 Fixing long lines in apprtc.py. 
These long lines causes the presubmit to get angry.

BUG=webrtc:2678
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5193 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-28 17:45:08 +00:00
pbos@webrtc.org
e1fc3f22ea Disable check for all sent SSRCs being valid. 
Since the code for setting these up will set the codec before setting
SSRCs for the streams, any frames sent in between will be sent on
random-generated SSRCs.

This part should be added back during work on issue 1695.

BUG=1695
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5192 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-28 15:40:12 +00:00
bjornv@webrtc.org
bd41a84694 This CL adds an API to enable robust validation of delay estimates. 
Added is
- a member variable for turning robust validation on and off.
- API to enable/disable feature.
- API to check if enabled.
- unit tests for these APIs.

Not added is
- the actual functionality (separate CL), hence turning feature on/off has no impact currently.
- calls in AEC and AEC, where the delay estimator is used. This is also done in a separate CL when we know if it should be turned on in both components.

TESTED=trybots, module_unittest
BUG=
R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4609005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5191 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-28 14:58:35 +00:00
stefan@webrtc.org
b627f676b3 Fixes a crash in the pacer where it fails to find a normal prio packet if there are no high prio packets, given that the queue has grown too large. 
BUG=2682
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4599005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5190 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-28 14:00:09 +00:00
pbos@webrtc.org
1f7c8d8b6a Lock frame in ViECapturer::IncomingFrameI420. 
r5160 explicitly assumed that IncomingFrameI420 was never called
sequentially. This assumption was found to be incorrect when some users
were changing beween existing capturers.

BUG=2657
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5189 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-28 13:26:33 +00:00
pbos@webrtc.org
13d38a13e3 Set up SSRCs correctly after switching codec. 
Before SSRCs were not set up correctly, as the old VideoEngine API
doesn't support setting additional SSRCs before a codec with as many
streams are set.

No test was in place to catch this, so two tests are added to make sure
that we send the SSRCs that are set, and also that we can switch from
using one to using all SSRCs, even though initially not all of them are
set up.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5188 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-28 11:59:31 +00:00
bjornv@webrtc.org
d1a1c353ac Recommit CL5184 
TBR=aluebs@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/4599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5187 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-28 11:45:05 +00:00
solenberg@webrtc.org
c8f76ddc19 Refactor Remote Estimators Test into a more reusable form. 
BUG=
R=andresp@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5186 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-28 10:11:35 +00:00
bjornv@webrtc.org
82eb3a690e Revert 5184 "Small refactoring change in delay_estimator." 
> Small refactoring change in delay_estimator.
> 
> This CL produce the bit exact output and is a preparing step for an upcoming robust validation scheme.
> 
> TESTED=trybots, module_unittest
> BUG=None
> R=aluebs@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/4549004

TBR=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5185 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-28 09:44:47 +00:00
bjornv@webrtc.org
eea079a376 Small refactoring change in delay_estimator. 
This CL produce the bit exact output and is a preparing step for an upcoming robust validation scheme.

TESTED=trybots, module_unittest
BUG=None
R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5184 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-28 07:59:04 +00:00
stefan@webrtc.org
19a40ff05b Ensure that no packet stays in the pacer queue for longer than 2 seconds. 
BUG=2682
TEST=trybots
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5182 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-27 14:16:20 +00:00
sprang@webrtc.org
b3ea3afa60 Create default implementation to fix issue in libjingle 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5181 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-27 10:28:27 +00:00
sprang@webrtc.org
4070935f4f Implement and test EncodedImageCallback in new ViE API. 
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5179 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-26 11:41:59 +00:00
asapersson@webrtc.org
c7ff8f990a Added measure of encode time. Added encode time to the ViE CpuOveruseMeasure api. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5178 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-26 11:12:33 +00:00
kjellander@webrtc.org
bd51d9324e LSan suppressions for libjingle_peerconnection_unittest 
Some more leaks have entered the code a while ago so
I'm suppressing them to green up the bot again.

BUG=2528
TEST=Successful execution of the test using the steps described in webrtc:2528.
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5177 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-26 09:37:12 +00:00
sprang@webrtc.org
7f959980f8 Remove const in vie_rtp_rtcp, where there is conflict with 
mock defines in fakewebrtcvideoengine.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5176 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-26 09:30:04 +00:00
mikhal@webrtc.org
d89b52af80 Faster implementation of BitRateStats. 
Landing cl for hguihot.

At high bitrate, EraseOld() could account for a significant part of
the total CPU usage on certain platforms. The new implementation
eliminates per-packet memory allocations and records the number of
bytes in buckets (one bucket per millisecond in window).

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5175 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-25 17:49:28 +00:00
elham@webrtc.org
326bcff879 Updated WebRTC version to 3.47 
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4349005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5173 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-25 17:19:46 +00:00
pbos@webrtc.org
4e3161dd55 Style-option file for clang-format. 
Specifies that clang-format as well as related editor plugins should use
Chromium style.

BUG=
TEST=formatting code with vim, running :ClangFormat before/after this change.
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5172 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-25 15:30:37 +00:00
phoglund@webrtc.org
3260f109e3 Made video quality toolchain more configurable. 
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4139007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5171 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-25 14:10:20 +00:00
stefan@webrtc.org
47fadba750 Add include stdlib.h to files using abs. 
abs function is declared in stdlib.h

Committing for alextaran@chromium.org.
Reviewed here: http://review.webrtc.org/4239004/

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5170 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-25 12:03:56 +00:00
stefan@webrtc.org
4ab4fc0044 Add test for automatically disabling padding when no video is being captured. 
BUG=2648
TEST=trybots
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5169 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-25 11:54:24 +00:00
fbarchard@google.com
b5bc098e20 Clear empty video frames in unittest so DrMemory will allow them to be read without an uninitialized read error. 
BUG=libyuv:263
TESTED=drmemory out\Debug\modules_unittests.exe --gtest_filter=*PreprocessorLogic
R=kjellander@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5168 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-25 09:06:33 +00:00
wu@webrtc.org
aa74b5d690 Add success/error callback to set sdp calls. 
Add a workaround for crbug/322756 to append a line break to the end of sdp if needed.

R=juberti@webrtc.org, vikasmarwaha@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5167 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-23 00:37:50 +00:00
turaj@webrtc.org
5272eb8d83 Don't register iSAC-swb and iSAC-fb in NetEqDecodingTest. 
Android bots break due to r5164. This CL patches that issue.

BUG=
TEST=modules_unittests on local device.
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5166 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-23 00:11:32 +00:00
sergeyu@chromium.org
e839da02c1 Fix MouseCursor to MouseCursorShape conversion in ScreenCapturerWin. 
BUG=crbug.com/322596
R=dcaiafa@chromium.org, wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/4279005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5165 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-22 20:39:16 +00:00
turaj@webrtc.org
78b41a09bd Fix issues with sequence number wrap-around in jitter statistics. 
Related CL for NetEq 3 is https://code.google.com/p/webrtc/source/detail?r=5150

Jitter statistics was not very sensitive to timestamp warp-around, and NetEqDecodingTest.TimestampWrap *DID NOT* fail before fixes applied. However, we still keep the test.

The criteria for the tests are not satisfied for first few packets, before any wrap-around happens. We could either relax the bound or ignore the first few packets. We chose the latter.

BUG=2662
TEST=modules_unittests,trybots
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5164 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-22 20:27:07 +00:00
fbarchard@google.com
832bd74cfa libyuv r874 for build improvements on ios/android, and improved YUV scale performance. 
BUG=libyuv:288
TESTED=try bots
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4229005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5163 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-22 19:30:52 +00:00
wu@webrtc.org
b43202d839 Disable PeerConnectionEndToEndTest for tsanv2 build. 
BUG=1205
TEST=try
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5162 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-22 19:14:25 +00:00
turaj@webrtc.org
1e8c93c953 Distinguish instances of ACM1 from ACM2 by a version string. This is fpr testing purposes and will be removed when the experiment is done and ACM1 is fade out. 
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4069006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5161 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-22 17:04:49 +00:00
pbos@webrtc.org
2ffb149c2c Replace VideoFrameI420 with I420VideoFrame. 
Gives one less struct/class for I420 video frames.

BUG=2657
R=mflodman@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5160 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-22 13:10:13 +00:00
andrew@webrtc.org
b0ed8f8a08 Don't reset the AEC filter in extended mode. 
I don't believe I've witnessed this "feature" ever provide a benefit,
and have now collected some evidence of its harm when using the
extended filter mode. It can cause erroneous resets in two cases:

1. Some preprocessing noise suppression is enabled in the system (i.e.
"audio enhancements") that push the noise floor very low, possibly to
zero. If the filter is non-zero this condition can be triggered very
easily, and erroneously.

2. Non-zero energy in the filter before the peak impulse response can
cause a slight (and harmless) "pre-echo" in the error signal. This
becomes more significant as the peak is set further back in the filter.
This effect can cause needless resets during echo onsets.

In short, this isn't a great criterion for filter reset and has the
potential to cause serious harm. Ideally we would remove it entirely,
but in the interests of safety, can start with the extended mode.

BUG=1261
R=aluebs@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5159 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-22 06:39:42 +00:00
dwkang@webrtc.org
9e85c01ec8 Add exception handling when configuring MediaCodc in order to prevent break in the new sdk release. 
BUG=2603
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5158 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-22 02:49:17 +00:00
henrik.lundin@webrtc.org
9fe3603dc1 Renaming ViEEncoderObserver::VideoSuspended 
New name is ViEEncoderObserver::SuspendChange.

BUG=2436
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5157 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 23:00:40 +00:00
pbos@webrtc.org
484ee962b5 Protect reads of ViEEncoder::video_suspended_. 
Does not fix an immediate bug, since this is the only method writing to
it there are no concurrent writes, but this should be more future-proof
by protecting all accesses.

BUG=2606
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4109006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5156 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 18:44:23 +00:00
fischman@webrtc.org
1977960866 AppRTCDemo(ios): remove codesigning hack now that gyp signs by default. 
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4119005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5155 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 16:48:51 +00:00
stefan@webrtc.org
ef2d55461b Increase size of pacer window to 500 ms as that better matches the encoder. 
BUG=1812
R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4129006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5154 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 14:37:11 +00:00
henrik.lundin@webrtc.org
331d4402fc Connect pacer/padding to SuspendBelowMinBitrate 
The suspend function must not be engaged unless padding is also enabled.
This CL makes the connection so that the pacer and padding is enabled
when SuspendBelowMinBitrate is.

Had to change the unit test to make it aware of the padding packets.

BUG=2606
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5153 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 14:05:40 +00:00
pbos@webrtc.org
ffe1b17b57 Lock access to ModuleRtpRtcpImpl::simulcast_. 
Fixes race between RegisterSendPayload and SendOutgoingData.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4099006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5152 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 13:53:13 +00:00
pbos@webrtc.org
2c46f8d854 Rename DestroyStream methods to include Video. 
Matches r5135 which renames CreateSendStream->CreateVideoSendStream for
instance.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4109005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5151 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 13:49:43 +00:00
henrik.lundin@webrtc.org
6f6ba6edee Fix issues with sequence number wrap-around in jitter statistics 
Wrap-arounds in sequence numbers (and in timestamps) were not always
treated correctly. This is fixed by introducing two helper functions
for correct comparisons, and by casting to the right word size.

Also added a new member variable to AutomodeInst_t. The new member keeps
track of when the first packet has been registered in the automode code.
This was previously done implicitly (and not very good) using the fact
that the lastSeqNo and lastTimestamp members were initialized to zero.

Two new unit tests were added as part of this CL.
NetEqDecodingTest.SequenceNumberWrap was failing before the fixes were
made; now it is ok.

BUG=2654
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5150 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 13:17:29 +00:00
pbos@webrtc.org
b3cc78de28 Add -Wnon-virtual-dtor warning for C++ code. 
BUG=2659
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4119006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5149 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 11:42:02 +00:00
sprang@webrtc.org
72964bd4fb Make interface destructor virtual 
In summary, do this:

-  ~FrameCountObserver() {}
+  virtual ~FrameCountObserver() {}

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4099005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5148 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 09:09:54 +00:00
asapersson@webrtc.org
8d02f5dc71 Added API for enabling/disabling RTCP Receiver Reference Time extension. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3419005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5147 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 08:57:04 +00:00
asapersson@webrtc.org
54a05518e2 Increase run-time for full stack test for the rtt to be added reliably to the delay measurement. 
BUG=2592
R=holmer@google.com, phoglund@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5146 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 07:45:08 +00:00
braveyao@webrtc.org
425e1d0fb9 Typo in vie_autotest_win.cc 
BUG=2637
TEST=AutoTest
R=mflodman@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5145 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 02:17:01 +00:00