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2389 Commits

Author SHA1 Message Date
stefan@webrtc.org
8b94e3da0f Fix issue where padding is sent before media with undefined timestamps if not abs-send-time is enabled. 
This broke bandwidth estimation for calls without abs-send-time is enabled, but where RTX was.

BUG=
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6719 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-17 16:10:14 +00:00
aluebs@webrtc.org
4065988108 Remove unused ExperimentalNS API in AudioProcessing 
R=andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6718 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-17 11:32:09 +00:00
kwiberg@webrtc.org
2b6bc8d84f AudioBuffer: Eliminate the SplitChannelBuffer class 
It's just a container for two IFChannelBuffers, and doesn't earn its
keep. The main problem is that the number of methods it needs that
just forward calls to either of its two IFChannelBuffers was already
large, and was about to grow.

R=aluebs@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6717 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-17 09:46:37 +00:00
aluebs@webrtc.org
2561d52460 Simplify AudioBuffer::mixed_low_pass_data API 
R=andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6715 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-17 08:27:39 +00:00
kwiberg@webrtc.org
af93fc08a1 AudioBuffer: Let ChannelBuffer handle bounds checking of channel parameter 
R=aluebs@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6714 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-17 08:18:33 +00:00
kwiberg@webrtc.org
2ade42bd96 Add unit test for MediaFile WAV file writing 
R=aluebs@webrtc.org, andrew@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6713 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-17 08:11:32 +00:00
tkchin@webrtc.org
4a472fb18d Fixes up rtc so that it compiles on iOS 8 SDK. 
Adds support for UIInterfaceOrientationUnknown (new with in SDK) and makes it the same as
UIInterfaceOrientationPortrait.

R=noahric@google.com, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13029004

Patch from David Maclachlan <dmaclach@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6712 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-17 00:21:59 +00:00
minyue@webrtc.org
c56ae63ea6 r6709 lacks a change in BUILD.gn 
BUG=
R=marpan@google.com, marpan@webrtc.org, pbos@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6710 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-16 22:18:49 +00:00
minyue@webrtc.org
74aaf29a0f Raw packet loss rate reported by RTP_RTCP module may vary too drastically over time. This CL is to add a filter to the value in VoE before lending it to audio coding module. 
The filter is an exponential filter borrowed from video coding module.

The method is written in a new class called PacketLossProtector (not sure if the name is nice), which can be used in the future for more sophisticated logic.

BUG=
R=henrika@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6709 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-16 21:28:26 +00:00
tkchin@webrtc.org
2e3c97ddf5 Compile-time guard for iOS7 specific property. 
BUG=3487
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6706 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-16 19:59:05 +00:00
stefan@webrtc.org
4070b1db53 Print an info log instead of return an error if an external encoder is de-registered, but no corresponding internal encoder can be registered automatically. 
This is not an error case if for instance an external h.264 encoder is registered, but no internal implementation exists.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6704 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-16 11:20:40 +00:00
pbos@webrtc.org
63c60ed224 Remove old padding path in RTPSender. 
Removing RTPSender::SendPaddingAccordingToBitrate() as well as a couple
of arguments from SendPadData().

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6703 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-16 09:37:29 +00:00
kwiberg@webrtc.org
efb81d8d1f int16<->float conversions: Use size_t for array length argument, not int 
size_t is more appropriate for array lengths, since int might
theoretically be too small for a really large array. But more
importantly, if the caller's value is naturally of type size_t and the
function requires an int, VC++ will trigger warning C4267
(http://msdn.microsoft.com/en-us/library/6kck0s93.aspx) because the
implicit cast might be lossy, forcing the caller to do a manual cast.
Typing the function with size_t in the first place resolves the
problem.

R=aluebs@webrtc.org, andrew@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6702 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-16 08:36:52 +00:00
kwiberg@webrtc.org
0fa6366ed1 Define convenient FATAL_ERROR() and FATAL_ERROR_IF() macros 
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6701 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-16 08:34:58 +00:00
kwiberg@webrtc.org
e8ea33ccb1 nrsh1 is written before tmp321 is read, so needs to be earlyclobber 
Otherwise, the compiler is allowed to put them in the same register
under the assumption that all inputs are read before any
(non-earlyclobber) output is written, which in this case would result
in nrsh2 being corrupted.

BUG=3439
R=aluebs@webrtc.org, ljubomir.papuga@gmail.com

Review URL: https://webrtc-codereview.appspot.com/16089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6700 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-16 08:26:48 +00:00
jiayl@webrtc.org
bac5f0fb56 Fix an invalid memory access due to typo in win/cursor.cc. 
BUG=crbug/391468
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/19949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6698 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 20:32:03 +00:00
tkchin@webrtc.org
122caa51b1 After an audio interruption the audio unit no longer invokes its render callback, which results in a loss of audio. Restarting the audio unit post interruption fixes the issue. 
CL also replaces deprecated AudioSession calls with equivalent AVAudioSession ones.

BUG=3487
R=glaznev@webrtc.org, noahric@chromium.org

Review URL: https://webrtc-codereview.appspot.com/21769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6697 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 20:20:47 +00:00
tkchin@webrtc.org
42fe4350fe Remove Thread::RunningForChannelManager(). 
I haven't heard of this failing, so it should be safe to remove. Let me know if this isn't the case.

BUG=3388
R=andrew@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6695 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 17:52:43 +00:00
stefan@webrtc.org
89fd1e8e99 Improvements to the pacer where it lost some budget due to truncation errors. 
With this CL the resolution is increased to microseconds and proper rounding
is done in the Process() function. This means that we will be allowed to send
more than prior to r6664 as we previously truncated away parts of our budget.

We will also not lose budget due to inaccurate calculations in
TimeUntilNextProcess(), which was a regression in r6664.

BUG=cr/393950
TEST=out/Debug/webrtc_perf_tests --gtest_filter=RampUpTest.Simulcast
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6694 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 16:40:38 +00:00
pbos@webrtc.org
376b4ea93f Fix breakage introduced by r6691. 
ModuleRtpRtcpImpl returned incorrectly on RemoteNTP as the
RTCPReceiver::NTP changed return type.

BUG=
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6693 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 15:51:33 +00:00
pbos@webrtc.org
2f4b14e3f3 Make RTCP sender report send media bytes. 
r6654 changed RtpSender::Bytes() to return the number of bytes sent
instead of number of media bytes. This is used by VideoEngine for stats.
This change broke RTCP which sends this same count as the number of
payload bytes sent (excluding headers and padding).

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6691 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 15:25:39 +00:00
kwiberg@webrtc.org
ffa8dcab1e Eliminate unnecessary #include 
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6690 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 12:50:13 +00:00
kwiberg@webrtc.org
324f63ca38 rtc::Fatal output: Print space between # and message 
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6689 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 11:41:05 +00:00
pbos@webrtc.org
bc73871251 Remove the VPM denoiser. 
The VPM denoiser give bad results, is slow and has not been used in
practice. Instead we use the VP8 denoiser. Testing this denoiser takes
up a lot of runtime on linux_memcheck (about 4 minutes) which we can do
without.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6688 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 09:50:40 +00:00
henrike@webrtc.org
92a9bacf9a Rebase webrtc/base with r6682 version of talk/base: 
cls ported: r6671, r6672, r6679 (reverts and unreverts in r6680, r6682).
svn diff -r 6656:6682 http://webrtc.googlecode.com/svn/trunk/talk/base >
6682.diff
sed -i.bak "s/talk_base/rtc/g" 6682.diff
sed -i.bak "s/#ifdef WIN32/#if defined(WEBRTC_WIN)/g" 6682.diff
sed -i.bak "s/#if defined(WIN32)/#if defined(WEBRTC_WIN)/g" 6682.diff
patch -p0 -i 6682.diff

BUG=3379
TBR=tommi@webrtc.org,jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6683 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-14 22:03:57 +00:00
glaznev@webrtc.org
a4da771914 Fix deadlock in Android stopCapture() call. 
BUG=3467
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6673 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-14 17:01:53 +00:00
kjellander@webrtc.org
9bef551ba1 GN: Fix include paths for WebRTC in Chromium build. 
Most WebRTC source files are using full paths for includes which
requires the root to be in the include path.

This is currently handled in the common_inherited_config config in
webrtc/BUILD.gn: the .. include_dir.

However, when built from Chromium, the include
paths are not inherited in the same way when building the all target.
Building the 'webrtc' target of Chrome works without the changes
in this CL, but the default target fails.

BUG=3441
TEST=Built the default target from a Chromium checkout with
https://codereview.chromium.org/321313006/ applied and
src/third_party/webrtc linked to the webrtc folder of the WebRTC
workspace.

R=brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/15989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6670 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-13 09:02:54 +00:00
tommi@webrtc.org
9e1acc8728 Fix bugs introduced by https://code.google.com/p/webrtc/source/detail?r=6667 . 
A few places were relying on temporalIdx being signed. Fix to explicitly check
for kNoTemporalIdx.

TBR=pbos,stefan

Review URL: https://webrtc-codereview.appspot.com/13939005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6669 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-11 20:33:39 +00:00
tommi@webrtc.org
dd6780d85d Remove always-true expression. 
TBR=pbos

Review URL: https://webrtc-codereview.appspot.com/16059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6668 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-11 19:34:54 +00:00
tommi@webrtc.org
eec6ecdb1e Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC. 
---

Fixes for re-enabling more MSVC level 4 warnings: webrtc/ edition

This contains fixes for the following sorts of issues:
* Possibly-uninitialized local variable
* Signedness mismatch
* Assignment inside conditional

This also contains a small number of other cleanups to nearby code. In
particular several warning-disables for MSVC are removed because they don't seem
to be necessary (either that warning is not enabled or the code does not trigger
it).

BUG=crbug.com/81439
TEST=none
R=henrika@webrtc.org, pkasting@chromium.org

Review URL: https://webrtc-codereview.appspot.com/18769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6667 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-11 19:09:59 +00:00
pbos@webrtc.org
180e516bef Thread annotate RTCPSender. 
Also fixes data races in RTCPSender::SetCSRCStatus() and
RTCPSender::SetStartTimestamp().

BUG=
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6666 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-11 15:36:26 +00:00
stefan@webrtc.org
168f23faa5 Move pacer to fully use webrtc::Clock instead of webrtc::TickTime. 
This required rewriting the send-side delay stats api to be callback based, as otherwise the SuspendBelowMinBitrate test started flaking much more frequently since it had lock order inversion problems.

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

Review URL: https://webrtc-codereview.appspot.com/21869005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6664 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-11 13:44:02 +00:00
pbos@webrtc.org
a1bfcad3a3 Cast payload types to int for logging. 
uint8_t gets interpreted as char and printed as such, instead of being
printed in decimal, casting them to int allows us to read what payload
types are actually used without converting them from ASCII first.

BUG=chromium:390874
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6662 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-11 12:33:45 +00:00
aluebs@webrtc.org
fb2e7c22a0 Document that channels are stored contiguously in AudioBuffer 
R=andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6661 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-11 11:40:48 +00:00
tommi@webrtc.org
d212ffcfc6 Remove unnecessary build message. 
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6660 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-11 11:15:35 +00:00
stefan@webrtc.org
4ef438e2de Remove the send-side cname getter APIs from voice and video engine. 
These APIs aren't being used, and introduces deadlocks when using GetStats() in the new Call api. Having getters for cname at the send-side is pointless, as it's always the user who sets the cname.

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

Review URL: https://webrtc-codereview.appspot.com/16899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6659 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-11 09:55:30 +00:00
henrike@webrtc.org
b614d0626f Rebase webrtc/base with r6655 version of talk/base: 
cls to port: r6633,r6639 (there is no cl in between that affects base and all other talk/base cls took care of webrtc/base as well (see r6569, r6624)):
svn diff -r 6632:6639 http://webrtc.googlecode.com/svn/trunk/talk/base > 6655.diff
sed -i.bak "s/talk_base/rtc/g" 6655.diff
patch -p0 -i 6555.diff

BUG=3379
TBR=tommi@webrtc.org,jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6656 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 22:47:02 +00:00
pbos@webrtc.org
72491b9a90 Count total bytes sent in RTPSender::Bytes(). 
Previously only media bytes were included, this adds header bytes and
padding bytes to the calculation.

BUG=
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6654 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 16:24:54 +00:00
pbos@webrtc.org
0422100818 Fix data race in VCMTiming::ResetDecodeTime. 
Also thread annotating class.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6653 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 15:25:37 +00:00
pbos@webrtc.org
bd9c0920ec Skip encoding in fake VP8 encoder. 
Broke memcheck, FakeEncoder::Encode doesn't produce valid VP8 frames.

BUG=3424
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6652 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 13:21:40 +00:00
andresp@webrtc.org
7ae9108b60 Remove more unused tsan suppressions and fix call test passing the same decoder to multiple received streams. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6651 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 10:35:12 +00:00
pbos@webrtc.org
91f1752f2d Support VP8 encoder settings in VideoSendStream. 
Stop-gap solution to support VP8 codec settings in the new API until
encoder settings can be passed on to the VideoEncoder without requiring
explicit support for the codec.

BUG=3424
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6650 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 10:13:37 +00:00
andresp@webrtc.org
8f1512140e Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module into vie_channel. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6649 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 09:39:23 +00:00
bjornv@webrtc.org
5bde66e913 audio_processing: Updates aec_core_sse2.c with changes made to aec_common.h 
The change of definitions moved to aec_common.h was done in CL17839005.

BUG=3131
TBR=kwiberg@webrtc.org
TESTED=builds locally

Review URL: https://webrtc-codereview.appspot.com/16859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6648 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 08:09:50 +00:00
bjornv@webrtc.org
555fc78f27 Neon version of SubbandCoherence() 
The performance gain on a Nexus 7 reported by audioproc is ~1.4%

The output is NOT bit exact.  Any difference seen is +-1.

BUG=3131
R=bjornv@webrtc.org, cd@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17839005

Patch from Scott LaVarnway <slavarnw@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6647 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 08:03:11 +00:00
bjornv@webrtc.org
ac800c8004 Neon version of rftbsub_128() 
The performance gain on a Nexus 7 reported by audioproc is ~4.5%

The output is bit exact.

BUG=3131
TESTED=trybots and manually
R=bjornv@webrtc.org, cd@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19919005

Patch from Scott LaVarnway <slavarnw@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6646 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 07:53:13 +00:00
andresp@webrtc.org
5ac876bae0 Revert "Remove remains of WEBRTC_NO_STL." (rev 6641). 
Reason breaks linux_memcheck.

TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6645 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-10 07:41:59 +00:00
andresp@webrtc.org
47d1c98a4e Remove remains of WEBRTC_NO_STL. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6641 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-09 20:18:28 +00:00
jiayl@webrtc.org
10ef8fe611 Create FullScreenChromeWindowDetector in DesktopConfigurationOptions::CreateDefault. 
BUG=crbug/385294
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/18759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6640 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-09 19:41:32 +00:00
stefan@webrtc.org
7af12be781 Thread annotations for vie_encoder.cc/.h 
Review URL: https://webrtc-codereview.appspot.com/8739005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6638 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-09 14:46:31 +00:00