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Author SHA1 Message Date
wuchengli@chromium.org
89e8ffb395 Revert "Add support of texture frames for video capturer." 
This reverts commit 83c89cd003be75d7d06ef9a2b139588f08d280ca.

Reason: The Buildbot has detected a new failure on builder
Android Chromium-APK Tests.

BUG=chromium:362437
TBR=fischman@webrtc.org, perkj@webrtc.org, stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6253 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-27 14:12:58 +00:00
wuchengli@chromium.org
efe15355ee Add support of texture frames for video capturer. 
- Add ViECapturer unittest.
- Add CloneFrame function in I420VideoFrame.
- Encoders do not support texture yet and texture frames
  are dropped in ViEEncoder for now.

Corresponding CLs:
https://codereview.chromium.org/277943002
http://cl/66620352

BUG=chromium:362437
TEST=WebRTC video stream forwarding. Run video_engine_core_unittests and common_video_unittests.
R=fischman@webrtc.org, perkj@webrtc.org, stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6252 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-27 12:40:27 +00:00
henrik.lundin@webrtc.org
59336e85fb Adding R/W lock to SimulatedClock 
This change makes the SimulatedClock class thread safe.

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

Review URL: https://webrtc-codereview.appspot.com/15569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6251 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-27 09:34:58 +00:00
phoglund@webrtc.org
f666ecc60d Disabling flaky libjingle tests after fixit week. 
BUG=webrtc:3316,webrtc:3317,webrtc:3318
TBR=fischman@google.com

Review URL: https://webrtc-codereview.appspot.com/12569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6250 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-27 08:08:00 +00:00
asapersson@webrtc.org
ab6bf4f54c Added api for getting cpu measures using a struct. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6249 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-27 07:43:15 +00:00
henrik.lundin@webrtc.org
74767401f2 Fix a bug preventing FilePlayer from playing encoded wav files 
A bug in ACM2 prevented decoding and playout of wav files where the
audio data was encoded (i.e., not just linear PCM 16 bit data).

This CL fixes the issue, and adds a unit test for the FilePlayer.

BUG=3386
R=henrike@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21499005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6248 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-26 13:37:45 +00:00
asapersson@webrtc.org
1457b4737a First incoming packet was not accounted for in receive stats. Changed call order for incoming packet to receive statistics class. 
Receive stats is reset if the payload type changes. Update stats after a possible reset.

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

Review URL: https://webrtc-codereview.appspot.com/12549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6247 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-26 13:06:04 +00:00
buildbot@webrtc.org
727ff69829 (Auto)update libjingle 67872893-> 67873348 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6244 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 23:20:53 +00:00
buildbot@webrtc.org
75cb3dc5f2 (Auto)update libjingle 67869540-> 67872893 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6243 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 23:13:35 +00:00
mallinath@webrtc.org
b445f26f24 Fixing correct UMA metric for PeerConnection enabled with IPv4 Vs IPv6. 
BUG=N/A
TBR=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21499007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6242 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 22:19:37 +00:00
fischman@webrtc.org
440e1d1053 vie_autotest_android.cc: stop referring to undefined functions. 
The roll in r6240 exposed the fact that vie_autotest_android.cc has been
depending on vie_autotest_network.cc since forever, even though that file isn't
part of the build!  #if'ing the references out to green the build.

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17599005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6241 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 21:40:45 +00:00
fischman@webrtc.org
4610f1d427 Roll chromium_revision 266514:272489 
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6240 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 20:56:12 +00:00
henrike@webrtc.org
ddc79d0418 Rebase webrtc/base with r6232: 
cd webrtc/base
svn diff -r 6231:6232 http://webrtc.googlecode.com/svn/trunk/talk/base > 6232.diff
patch -p0 -i 6232.diff

BUG=3379
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6239 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 18:40:46 +00:00
fischman@webrtc.org
39eccefbde Disable ChannelManagerTest.StartupShutdownOnUnstartedThread 
The test is testing a scenario that shouldn't happen.

BUG=3388
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21509005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6238 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 17:50:38 +00:00
buildbot@webrtc.org
7aa1a4767f (Auto)update libjingle 67848628-> 67848776 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6237 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 17:33:05 +00:00
fischman@webrtc.org
e5063b1733 Thread: delete racy API (Release()) and fix racy code (started()). 
- Thread::Release() wrote a local variable on the calling thread but read it on
  another thread, with no synchronization.  Happily it has no non-test callers
  so deleting it instead of trying to fix it (see bug for details).
- Thread::started_ similarly was racily being written to; replaced with a
  running_ Event, and hid the accessor except for tests & legacy callers,
  with a note about why it's a bad idea.

webrtc/base patched with:
git diff origin --relative=talk/base | patch -p1 -dwebrtc/base
followed by manual merge of 3 thunks that ran afoul of naming differences
between talk/base and webrtc/base.

BUG=3388
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14589005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6236 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 17:28:50 +00:00
fischman@webrtc.org
18f41b8eb4 PRESUBMIT.py: accept variants on the copyright message that are present in the codebase. 
Example files that this makes ok instead of flagging include:
  talk/base/signalthread_unittest.cc
  talk/base/thread_unittest.cc
  webrtc/base/signalthread_unittest.cc
  webrtc/base/thread.cc
  webrtc/base/thread.h
  webrtc/base/thread_unittest.cc

BUG=1027
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19539006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6235 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 17:27:18 +00:00
turaj@webrtc.org
546961a9d3 Avoid reading uninitialized values (outside baundary) in DFT arithmatic decoder of iSAC-fix. 
Arithmetic encoder does not right the last 2 or 3 bytes of |streamval| when terminating the bit-stream. Perhaps the last bytes makes no difference in decoding the stream. However, the decoder reads full |streamval| (int16_t) going out of boundary and reading uninitialized values. This avoids this problem. by inserting zero-bytes whenever decoder intends to read outside boundary.

BUG=1353,chrome373312,b/13468260
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16499005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6234 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 17:14:29 +00:00
minyue@webrtc.org
aa5ea1c0f9 1. Make a clear distinction between codec internal FEC and RED, confusing mentioning of FEC in the old codes is replaced by RED 
2. Add two new APIs to configure codec internal FEC

3. Add a test and listened to results. This is based modifying EncodeDecodeTest and deriving a new class from it.

New ACM gives good result.
Old ACM does not use NetEq 4, so FEC won't be decoded.

BUG=
R=tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6233 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 15:16:51 +00:00
pbos@webrtc.org
706152dcc9 Fix uninitialized reads in IsDefaultBrowserFirefox 
BUG=
TEST=Local DrMemory.
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19529006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6232 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 14:35:48 +00:00
pbos@webrtc.org
1566ee2893 Revert "Revert "Remove VideoSendStreamInput::PutFrame."" 
This reverts commit r6230 to re-land r6229.

ViECapturer::SwapFrame now resets timestamps.

BUG=
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6231 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 13:03:45 +00:00
pbos@webrtc.org
2cdd433edf Revert "Remove VideoSendStreamInput::PutFrame." 
This reverts r6229.

Test WebRtcVideoChannel2BaseTest.MuteStream fails after r6229.

BUG=
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19529005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6230 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 10:43:26 +00:00
pbos@webrtc.org
f3085e43ab Remove VideoSendStreamInput::PutFrame. 
PutFrame just copies the frame before swapping it, if it's required that
can easily be done outside this API before swapping the frame.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14529006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6229 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 09:41:45 +00:00
pbos@webrtc.org
6e98ef4b35 Fix deadlock in RegisterPreDecodeImageCallback. 
Fixes lock-order inversion between ViEChannel::callback_cs_ and
VideoReceiver::_receiveCritSect detected on DrMemory Full which
exhibited different timing behavior.

Also removes most of the suppressions on DrMemory Full as they're able
to run again without deadlocking.

BUG=3336,3375
TEST=Run DrMemory Full trybots.
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6228 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 09:41:07 +00:00
pbos@webrtc.org
bc524ae41a Added mirror of gtest-parallel. 
gtest-parallel is a Python script that runs gtest binaries in parallel.

R=andrew@webrtc.org, kjellander@webrtc.org, mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/11309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6227 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 09:37:29 +00:00
stefan@webrtc.org
b60bfe4759 Suppress webrtc trace races detected by tsan. 
BUG=3372
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6226 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 07:29:37 +00:00
wu@webrtc.org
10f871f29b Remove the restriction to allow having both webrtc and talk changes in the same cl. 
This restriction is no longer needed as the auto sync script can handle changes to both folder in same commit correctly.

BUG=
R=andrew@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15479006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6225 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-22 22:35:46 +00:00
tnakamura@webrtc.org
0720758f9f Bump WebRTC version number to 3.54 
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17619006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6222 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-22 17:51:18 +00:00
henrike@webrtc.org
1bb5da04fe Adds missing include of assert header. 
BUG=3380
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/14569008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6221 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-22 14:31:14 +00:00
braveyao@webrtc.org
21f7d6d2fe WebRTCDemo: move the deletion of CritSect to end of the dtor to fix a crash in Android video renderer. 
BUG=3368
TEST=Manual Test

Review URL: https://webrtc-codereview.appspot.com/21519005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6220 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-22 02:57:55 +00:00
mallinath@webrtc.org
8e755c1ad2 Connect SignalDestroyed in AllocationSequence after TURN ports are destroyed 
when TURN ports are using shared socket with UDP port.

This is required as AllocationSequence maintains a map of turn ports. If the
ports are destroyed without the knowledge of AllocationSequence, sequence will
try to deliver packets to the destoyed ports.

R=jiayl@webrtc.org
BUG=https://code.google.com/p/chromium/issues/detail?id=368877

Review URL: https://webrtc-codereview.appspot.com/14569007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6219 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 23:00:46 +00:00
henrike@webrtc.org
88fbb2d86b Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. 
Same as https://webrtc-codereview.appspot.com/19519004. The issue in
http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Linux...
is solved by this change
http://src.chromium.org/viewvc/chrome/trunk/src/third_party/libjingle/libjing...
(tested locally).

BUG=3380
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17619005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6218 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 21:18:46 +00:00
henrike@webrtc.org
99b4162ccf Rebase webrtc/base 6163:6216 (svn diff -r 6163:6216 http://webrtc.googlecode.com/svn/trunk/talk/base, apply diff manually) 
BUG=3379
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6217 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 20:42:17 +00:00
buildbot@webrtc.org
f9f1bfbdae (Auto)update libjingle 67686255-> 67689476 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6216 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 17:02:15 +00:00
henrike@webrtc.org
a148704b4b Rename webrtc/base's IS_ALIGNED macro to RTC_IS_ALIGNED to avoid conflict between webrtc/base/basictypes.h and third_party/.../vpx_codec.h. 
BUG=3380
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17579005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6215 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 16:52:14 +00:00
buildbot@webrtc.org
ce4201df52 (Auto)update libjingle 67643194-> 67686255 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6214 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 16:22:51 +00:00
jiayl@webrtc.org
7ca277b574 Initializes WINDOWPLACEMENT::length in GetCroppedWindowRect. 
BUG=https://code.google.com/p/webrtc/issues/detail?id=3196
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/21529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6213 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 16:02:31 +00:00
henrike@webrtc.org
000658a138 Revert of 6211 as it was committed despite of PRESUBMIT.py warning. The commit breaks the sync bot. 
BUG=N/A
TBR=mcasas@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21519006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6212 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 16:01:13 +00:00
mcasas@webrtc.org
3b7e282caa Disabling systematically failing 
WebRtcVideoMediaChannelTest.SendVp8HdAndReceiveAdaptedVp8Vga

TBR= pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14569006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6211 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 14:25:20 +00:00
mcasas@webrtc.org
2fa7f79094 Revert 6202 "Switch to using base/constructormagic.h and remove ..." 
> Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
> 
> BUG=N/A
> R=andrew@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/19519004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14579007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6210 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 11:07:29 +00:00
mcasas@webrtc.org
c2213b6a0f Revert 6208 "Patch from henrike@webrtc.org" 
Wasn't enough. I'll have to revert the whole rev 6202.

> Patch from henrike@webrtc.org
> https://code.google.com/p/webrtc/source/detail?r=6202
> didn't work for at least one file and broke most of 
> the compile steps in the FYI bots. The file is reverted
> here.
> 
> TBR= henrike@webrtc.org, sergeyu@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/17609004

TBR=mcasas@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14579006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6209 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 10:03:09 +00:00
mcasas@webrtc.org
86df8acc92 Patch from henrike@webrtc.org 
https://code.google.com/p/webrtc/source/detail?r=6202
didn't work for at least one file and broke most of 
the compile steps in the FYI bots. The file is reverted
here.

TBR= henrike@webrtc.org, sergeyu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6208 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 08:40:56 +00:00
braveyao@webrtc.org
1a79bb8d30 WebRTCDemo: clean the error message due to API clean up and add ability to route the audio through all three outputs, headset/earpiece/loudspeaker 
BUG=3366
TEST=Manual Test
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6207 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 03:37:45 +00:00
buildbot@webrtc.org
49a6a27bf0 (Auto)update libjingle 67555838-> 67643194 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6206 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 00:24:54 +00:00
wu@webrtc.org
82c4b8531c Calculate capture ntp timestamp in local timebase for decoded audio frame. 
BUG=3111
R=stefan@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19449005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6205 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 22:55:01 +00:00
henrik.lundin@webrtc.org
48438c2c90 Enabling NetEq bit-exactness test for Win x64 
A new reference file (neteq4_universal_ref_win_64.pcm) was generated and
uploaded.

Also removing the old hack to have different reference files
for different version of Visual Studio. The test is now only supporting
VS 2012 and later (_MSC_VER >= 1700). This makes the windows 32-bit
output identical to the generic reference file
(neteq4_universal_ref.pcm), so the specialized one
(neteq4_universal_ref_win_32.pcm) could have been removed. However,
since the resources sync mechanism does not include removing of old
files, a client could pick up the old reference and fail. Therefore,
this cl also updates neteq4_universal_ref_win_32.pcm to be identical to
neteq4_universal_ref.pcm.

BUG=1458
R=kjellander@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14569005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6204 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 16:07:43 +00:00
henrik.lundin@webrtc.org
aed31fe8ab Modifying WATCHLISTS 
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6203 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 16:05:47 +00:00
henrike@webrtc.org
125ffd709d Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6202 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 15:20:44 +00:00
stefan@webrtc.org
4059c2f579 Disable CallPerfTest.CaptureNtpTimeWithNetworkDelay due to being flaky. 
TBR=wu@webrtc.org
BUG=3374

Review URL: https://webrtc-codereview.appspot.com/14579005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6201 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 15:12:29 +00:00
stefan@webrtc.org
70bb2d5755 Revert r6198 "Expose the original packet length in in the RTP play tools." 
TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6200 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 13:25:48 +00:00