Fix bugs in DesktopRegion::IntersectWith() and DesktopRect::IntersectWith().

IntersectWith() didn't work correctly which breaks screen capturers in chromium.

BUG=crbug.com/243160
R=alexeypa@chromium.org, wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1560004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4102 4adac7df-926f-26a2-2b94-8c16560cd09d

Remove dead testRateControl.cc

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1556004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4101 4adac7df-926f-26a2-2b94-8c16560cd09d

Removed dead testH263Parser.cc

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1555004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4100 4adac7df-926f-26a2-2b94-8c16560cd09d

Remove dead bitstreamTest.cc.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1553004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4099 4adac7df-926f-26a2-2b94-8c16560cd09d

Make sure GlxRenderer frees its resources.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1544004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4098 4adac7df-926f-26a2-2b94-8c16560cd09d

Adds integration test for RTX and fixes bugs found.

BUG=1811
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4096 4adac7df-926f-26a2-2b94-8c16560cd09d

Fix regression where retransmission bitrate is no longer estimated.

BUG=1813
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1530004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4095 4adac7df-926f-26a2-2b94-8c16560cd09d

CreateEmptyFrame casts from size_t to int.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1540004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4094 4adac7df-926f-26a2-2b94-8c16560cd09d

FrameGenerator class for future fake capture device.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1511004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4093 4adac7df-926f-26a2-2b94-8c16560cd09d

Control new VideoEngine tests with gflags.

BUG=1703
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1497005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4092 4adac7df-926f-26a2-2b94-8c16560cd09d

Adds print out of incoming resolution.

BUG=N/A
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1532004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4091 4adac7df-926f-26a2-2b94-8c16560cd09d

Log the type of recycled frames.

Also correct the logging of incoming key frame packets.

BUG=1814
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1537004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4090 4adac7df-926f-26a2-2b94-8c16560cd09d

Log a message when a key frame packet is received

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1518004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4089 4adac7df-926f-26a2-2b94-8c16560cd09d

Fix failing tests on 32 bit Linux.

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1534004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4088 4adac7df-926f-26a2-2b94-8c16560cd09d

API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay.

TEST=unit-test, manual, trybots.
R=henrik.lundin@webrtc.org, henrika@webrtc.org, mflodman@webrtc.org, mikhal@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1384005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4087 4adac7df-926f-26a2-2b94-8c16560cd09d

- Changed RemoteBitrateEstimator::IncomingPacket() to include a const WebRtcRTPHeader& and remove ssrc, rtp_timestamp.
- Changed RemoteBitrateObserver::OnReceivedBitrateChanged() to use a const & instead of non-const *, to avoid unnecessary copying.
- Refactored RemoteBitrateEstimatorTest so it can be instantiated for both single and multi stream BWE (first using a parameterized test, but then as a standard test fixture and a few helper functions).
- Refactored some tests in RemoteBitrateEstimatorTest into a common function CapacityDropTestHelper().

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

Review URL: https://webrtc-codereview.appspot.com/1521004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4086 4adac7df-926f-26a2-2b94-8c16560cd09d

Disable WindowCapturer tests on OSX and Linux

R=alexeypa@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1533004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4085 4adac7df-926f-26a2-2b94-8c16560cd09d

Add direct_dependent_settings in common.gypi.

When building chromium targets that depend on webrtc, compiler settings must
have the include path to webrtc and webrtc-specific defines that the headers
may depend on. Added direct_dependent_settings in common.gyp, so that all
webrtc target propagate these settings to dependencies.

R=andrew@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1371005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4084 4adac7df-926f-26a2-2b94-8c16560cd09d

Not to request to TURN server for local tests. Follow-up work to issue1197.

BUG=1197
TEST=Manual test
R=dutton@google.com

Review URL: https://webrtc-codereview.appspot.com/1340004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4083 4adac7df-926f-26a2-2b94-8c16560cd09d

Roll libvpx to 196669.
  -pick up libvpx roll to 9981006d

TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1523004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4082 4adac7df-926f-26a2-2b94-8c16560cd09d

Refactor VCM/Timing.
No changes in functionality.

R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/1514004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4081 4adac7df-926f-26a2-2b94-8c16560cd09d

Consolidate GetFrame and InsertPacket and move NACK list processing to after a packet has been successfully inserted.

TEST=trybots
BUG=1799
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4080 4adac7df-926f-26a2-2b94-8c16560cd09d

Include files from webrtc/.. paths in voice_engine/

BUG=1662
R=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1434005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4079 4adac7df-926f-26a2-2b94-8c16560cd09d

Make sure VoiceEngine tests only include one test framework.

BUG=
R=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4078 4adac7df-926f-26a2-2b94-8c16560cd09d

Remove <iostream> usage from loopback.cc

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1522004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4077 4adac7df-926f-26a2-2b94-8c16560cd09d

Suffix VcmCapturer's privates with underscore_

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1506005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4076 4adac7df-926f-26a2-2b94-8c16560cd09d

Log timestamp of the frame when it's dropped from the render module

R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1515005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4075 4adac7df-926f-26a2-2b94-8c16560cd09d

Log error in ViESender::SendRTCPPacket

Log the packet length and the error of SendRTCPPacket.

R=mikhal@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1512005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4074 4adac7df-926f-26a2-2b94-8c16560cd09d

Revert 4067 "libyuv roll to r698 for Core Media fourccs for OSX ..."

> libyuv roll to r698 for Core Media fourccs for OSX camtwist support and performance improvements in ARGB scaler.
> BUG=none
> TEST=libyuv unittests add CM32 and CM24 types and ARGBScaleClip tests added.
> Review URL: https://webrtc-codereview.appspot.com/1508004

TBR=fbarchard@google.com

Review URL: https://webrtc-codereview.appspot.com/1517004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4072 4adac7df-926f-26a2-2b94-8c16560cd09d

Revert 4000 "Reverting r3978"

> Reverting r3978
>
> BUG=webrtc:1749
> R=niklas.enbom@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1454004

TBR=elham@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1516004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4071 4adac7df-926f-26a2-2b94-8c16560cd09d

Revert 4001 "Revert 3977"

> Revert 3977
> BUG=webrtc:1749
>
> > Update protoc.gypi to match Chromium's latest.
> >
> > This is in preparation for enabling protobufs in Chromium. Requires
> > syncing tools/protoc_wrapper.
> >
> > BUG=webrtc:830
> > R=kjellander@webrtc.org
> >
> > Review URL: https://webrtc-codereview.appspot.com/1426004
>
> TBR=andrew@webrtc.org
> Review URL: https://webrtc-codereview.appspot.com/1453005

TBR=tnakamura@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1515004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4070 4adac7df-926f-26a2-2b94-8c16560cd09d

Fix assertions in rtp_header_extension.h caused by not handling the AudioLevel extension. Added unit tests to do basic checks of the AudioLevel extension.

BUG=
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1510004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4069 4adac7df-926f-26a2-2b94-8c16560cd09d

Recalibrate point sample expectation
BUG=none
TESTED=try bots
Review URL: https://webrtc-codereview.appspot.com/1512004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4068 4adac7df-926f-26a2-2b94-8c16560cd09d

libyuv roll to r698 for Core Media fourccs for OSX camtwist support and performance improvements in ARGB scaler.
BUG=none
TEST=libyuv unittests add CM32 and CM24 types and ARGBScaleClip tests added.
Review URL: https://webrtc-codereview.appspot.com/1508004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4067 4adac7df-926f-26a2-2b94-8c16560cd09d

Add functions to ViE API to enable/disable the absolute send time header extension.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1487004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4065 4adac7df-926f-26a2-2b94-8c16560cd09d

Window capturer implementation for Windows.

R=alexeypa@chromium.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1477004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4064 4adac7df-926f-26a2-2b94-8c16560cd09d

AppRTC: make requestTurn() failure non-fatal to call establishment.

BUG=1795
R=vikasmarwaha@google.com, vikasmarwaha@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1504005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4063 4adac7df-926f-26a2-2b94-8c16560cd09d

Avoid NPE crash on Android platforms that don't support getting preview framerate.
- catch Camera.setParameters() signaling errors through RuntimeException (!)
- make video_demo_apk rebuild when .java sources change

BUG=1778
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1493004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4059 4adac7df-926f-26a2-2b94-8c16560cd09d

Enable WebRTC demo application on x86 Android

Steps to build the demo application for x86 Android:
source build/android/envsetup.sh --target-arch=x86
gclient runhooks
ninja -C out/Debug
cd webrtc/video_engine/test/android
ndk-build APP_ABI=x86
ant debug

Commmitted as https://code.google.com/p/webrtc/source/detail?r=4053

R=fischman@webrtc.org, leozwang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1478004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4058 4adac7df-926f-26a2-2b94-8c16560cd09d

Include gflags properly and X11 include order in VideoEngine.

BUG=

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1500004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4057 4adac7df-926f-26a2-2b94-8c16560cd09d

Include files from webrtc/.. paths in video_engine/

BUG=1662
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1492004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4056 4adac7df-926f-26a2-2b94-8c16560cd09d

Improve wraparound handling in the render time extrapolator.

This was actually working as intended, but as r3970 changed when render timestamps were extrapolated to when a frame was taken out for decoding, the wraparound could have happened in the Update() step before it had happened in the ExtrapolateLocalTime() step. This causes render timestamps to be generated 13 hours into the future.

TEST=trybots
BUG=1787
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1497004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4055 4adac7df-926f-26a2-2b94-8c16560cd09d

Moved command line parsing to internal tools and moved back the mic volume thingie.

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

Review URL: https://webrtc-codereview.appspot.com/1491004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4054 4adac7df-926f-26a2-2b94-8c16560cd09d

Enable WebRTC demo application on x86 Android

Steps to build the demo application for x86 Android:
source build/android/envsetup.sh --target-arch=x86
gclient runhooks
ninja -C out/Debug
cd webrtc/video_engine/test/android
ndk-build APP_ABI=x86
ant debug

R=fischman@webrtc.org, leozwang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1478004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4053 4adac7df-926f-26a2-2b94-8c16560cd09d

Guarding certain operations, e.g. bandwidth estimation, RTCP statistics update etc., not to be run on sync RTPS.

BUG=issue1770
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1485004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4052 4adac7df-926f-26a2-2b94-8c16560cd09d

Add one unit test for NACKing a key frame

Adding a test case that wasn't covered. This new test is passing.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1475004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4051 4adac7df-926f-26a2-2b94-8c16560cd09d

Cleanup traces in WebRTC

Remove some unused traces and add a trace counter for encoded video size.

R=holmer@google.com, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1476004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4050 4adac7df-926f-26a2-2b94-8c16560cd09d

Avoid resetting encoder on identical settings.

BUG=1681
R=holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1481005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4049 4adac7df-926f-26a2-2b94-8c16560cd09d

Bugfix: VCM would report wrong sentBitrate

issue: https://code.google.com/p/webrtc/issues/detail?id=1755

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1484004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4048 4adac7df-926f-26a2-2b94-8c16560cd09d

Formatted FEC stuff.

Unfortunately I had to pull in quite a bit of stuff due to use of unencapsulated public member variables.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1401004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4047 4adac7df-926f-26a2-2b94-8c16560cd09d

Moved force_volume_max to its own gyp file to avoid a circular dependency.

BUG=
TBR=tlegrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4046 4adac7df-926f-26a2-2b94-8c16560cd09d

Wrote a small portable tool for forcing the mic volume to 100%.

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

Review URL: https://webrtc-codereview.appspot.com/1477005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4045 4adac7df-926f-26a2-2b94-8c16560cd09d

New VideoEngine API implementation on top of old one, first steps.

BUG=1668
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1360004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4044 4adac7df-926f-26a2-2b94-8c16560cd09d

Log too long non-decodable duration events.

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1488004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4043 4adac7df-926f-26a2-2b94-8c16560cd09d

Remove SetOverUseDetectorOptions and cleaned ViESharedData.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1486004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4042 4adac7df-926f-26a2-2b94-8c16560cd09d

Add handling of the absolute send time header extension to the rtp_rtcp module.

BUG=
R=asapersson@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1480004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4041 4adac7df-926f-26a2-2b94-8c16560cd09d

Updated apprtc demo to interop with firefox.

R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/1482004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4040 4adac7df-926f-26a2-2b94-8c16560cd09d

Added webaudio-and-webtrc.html to the demos index.html.

R=dutton@google.com, henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1425005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4039 4adac7df-926f-26a2-2b94-8c16560cd09d

Roll chromium_revision 193311:199267

This will fix static libraries will not be copied to product out dir issue on x86 Android

Remove third_party/WebKit/Tools/Scripts since it will not be used.

BUG=webrtc:1690
TEST=Trybots passing
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1457004

Patch from Jeremy Mao <yujie.mao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4038 4adac7df-926f-26a2-2b94-8c16560cd09d

Updating NACK RTX test

BUG=1513
R=holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1274006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4036 4adac7df-926f-26a2-2b94-8c16560cd09d

VCM/JB: Bug fix in ExtractAndSetDecode
BUG=1771
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1466005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4035 4adac7df-926f-26a2-2b94-8c16560cd09d

RemoteBitrateEstimatorTest::TestRateIncreaseReordering sent in arrival timestamps in non monotonically increasing order. Fixed.

BUG=
R=holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1481004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4034 4adac7df-926f-26a2-2b94-8c16560cd09d

CoreAudio Win: release resources safely under certain rare circumstance in GTalkplugin

BUG=
TEST=voe_auto_test
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4033 4adac7df-926f-26a2-2b94-8c16560cd09d

Linux support for typing detection

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1428006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4031 4adac7df-926f-26a2-2b94-8c16560cd09d

Address sanitizer out of bounds read in iSAC

BUG=issue1770
TBR=tlegrand@google.com

Review URL: https://webrtc-codereview.appspot.com/1472006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4030 4adac7df-926f-26a2-2b94-8c16560cd09d

Remove const for plain data types in common_video/

BUG=1644
R=holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1464004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4028 4adac7df-926f-26a2-2b94-8c16560cd09d

Adding a factory to remote bitrate estimator and allow it to be set via config.

Additionally:
 - clean api to set remote bitrate estimator mode.
 - clean api to set over use detector options.

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

Review URL: https://webrtc-codereview.appspot.com/1448006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4027 4adac7df-926f-26a2-2b94-8c16560cd09d

Fixes a bug where the render buffer size (and indirectly the non-continuous duration) was computed incorrectly.

BUG=1769
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1473004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4026 4adac7df-926f-26a2-2b94-8c16560cd09d

Fixed more perf expectations.

For Linux, the expectations just look a bit too tightly wound. On Windows there's a long-term increasing trend that we may want to have someone look at.

http://www.corp.google.com/~webrtc-cb/perf//linux-large-tests/vie_auto_test/report.html?history=1500&rev=-1&graph=total_delay_incl_network
http://www.corp.google.com/~webrtc-cb/perf//linux-large-tests/vie_auto_test/report.html?history=1500&rev=-1&graph=total_delay_incl_network

BUG=
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1472005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4025 4adac7df-926f-26a2-2b94-8c16560cd09d

Adjusted perf expectations for mac large tests.

BUG=
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1472004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4024 4adac7df-926f-26a2-2b94-8c16560cd09d

Removed Mac capture crash and memory leak.

BUG=1697,1761
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1465005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4023 4adac7df-926f-26a2-2b94-8c16560cd09d

Add script for comparing video quality

This script makes it easier to run a simple command line
comparison between a captured YUV file and a reference video.

BUG=none
TEST=command line invocation
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1320007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4022 4adac7df-926f-26a2-2b94-8c16560cd09d

Added protoc_wrapper to blacklist, fixed tools/PRESUBMIT.py which was passing in the wrong args to CheckLongLines.

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

Review URL: https://webrtc-codereview.appspot.com/1470005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4021 4adac7df-926f-26a2-2b94-8c16560cd09d

Reformatted FEC tables.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1400004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4020 4adac7df-926f-26a2-2b94-8c16560cd09d

Remove const for plain data types in common_audio/

BUG=1644
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1464005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4019 4adac7df-926f-26a2-2b94-8c16560cd09d

Remove const for plain data types in voice_engine/

BUG=1644
R=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1463004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4018 4adac7df-926f-26a2-2b94-8c16560cd09d

Replace ExtraCodecOptions with new Config class that supports multiple settings at once.

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1452004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4017 4adac7df-926f-26a2-2b94-8c16560cd09d

Fix typo in log statement.  witdh should be width.
BUG=none
TESTED=try bots
Review URL: https://webrtc-codereview.appspot.com/1466004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4016 4adac7df-926f-26a2-2b94-8c16560cd09d

Add more tracing for key frames.

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

Review URL: https://webrtc-codereview.appspot.com/1428004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4015 4adac7df-926f-26a2-2b94-8c16560cd09d

Increased the limit for KViEMaxCaptureDevices from 10 to 256. See issue 1343.
TBR=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/1463005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4014 4adac7df-926f-26a2-2b94-8c16560cd09d

Added Stereo url paramter to apprtc demo.

R=dutton@google.com

Review URL: https://webrtc-codereview.appspot.com/1418004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4013 4adac7df-926f-26a2-2b94-8c16560cd09d

Updated WebRTC version to 3.31
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1462004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4011 4adac7df-926f-26a2-2b94-8c16560cd09d

Revert 4008 "Avoid resetting video encoder for similar configs."

> Avoid resetting video encoder for similar configs.
>
> BUG=1681
> R=holmer@google.com, mflodman@webrtc.org, stefan@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1442006

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1431005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4010 4adac7df-926f-26a2-2b94-8c16560cd09d

Disabled flaky codec test (RunsCodecTestWithoutErrors)

BUG=1734
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1460004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4009 4adac7df-926f-26a2-2b94-8c16560cd09d

Avoid resetting video encoder for similar configs.

BUG=1681
R=holmer@google.com, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1442006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4008 4adac7df-926f-26a2-2b94-8c16560cd09d

Wiring down config from video engine until video coding and remote bitrate estimator modules instantiation.

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1450008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4007 4adac7df-926f-26a2-2b94-8c16560cd09d

New WebAudio-WebRTC demo.

Capture microphone input and stream it out to a peer with a processing effect applied to the audio.

The audio stream is:

o Recorded using live-audio input.
o Filtered using an HP filter with fc=1500 Hz.
o Encoded using Opus.
o Transmitted (in loopback) to remote peer using RTCPeerConnection where it is decoded.
o Finally, the received remote stream is used as source to an <audio> tag and played out locally.

Press any key to add an effect to the transmitted audio while talking.

Please note that:

o Linux is currently not supported.
o Sample rate and channel configuration must be the same for input and output sides on Windows.
o Only the Default microphone device can be used for capturing.

R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1256004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4006 4adac7df-926f-26a2-2b94-8c16560cd09d

Remove TEXT(x) for BUILDINFO macros.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1453004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4005 4adac7df-926f-26a2-2b94-8c16560cd09d

Added a config class to ease passing a set of options across webrtc.
Its main design reason is to expose control of experimental webrtc features.

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1450009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4004 4adac7df-926f-26a2-2b94-8c16560cd09d

Add svn:eol-style back which is lost in r3993 mistakenly.

R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1428008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4003 4adac7df-926f-26a2-2b94-8c16560cd09d

Change watchlist.

Watch all changes in webrtc.

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1428012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4002 4adac7df-926f-26a2-2b94-8c16560cd09d

Revert 3977
BUG=webrtc:1749

> Update protoc.gypi to match Chromium's latest.
>
> This is in preparation for enabling protobufs in Chromium. Requires
> syncing tools/protoc_wrapper.
>
> BUG=webrtc:830
> R=kjellander@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1426004

TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1453005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4001 4adac7df-926f-26a2-2b94-8c16560cd09d

Reverting r3978

BUG=webrtc:1749
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1454004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4000 4adac7df-926f-26a2-2b94-8c16560cd09d

This is the first step to convert building the Android WebRTC demo to a proper GYP target, android ndk toolchains is being used to build the jni cpp files instead of using ndk-build.

R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1444005

Patch from Jeremy Mao <yujie.mao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3999 4adac7df-926f-26a2-2b94-8c16560cd09d

Updating perf

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1428011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3997 4adac7df-926f-26a2-2b94-8c16560cd09d

Use 2 threads for HD, or 1 for VGA or less.
BUG=1739
TEST=try bots
Review URL: https://webrtc-codereview.appspot.com/1438005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3996 4adac7df-926f-26a2-2b94-8c16560cd09d

Updating perf

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1447004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3995 4adac7df-926f-26a2-2b94-8c16560cd09d

Since the layout of the Android WebRTC demo application is fixed, if we start the demo application in portrait postion, the activity will be destroyed and then created again, force the demo application to start in landscape position to avoid activity re-creation.

BUG=webrtc:1741

TEST=Build and run the Android WebRTC demo application
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1439006

Patch from Jeremy Mao <yujie.mao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3994 4adac7df-926f-26a2-2b94-8c16560cd09d

WebRTCDemo Android doesn't hangle activity recreation correctly.
Also optimize Statsview a little bit.

BUG=1740
TEST=Manual test with WebRTCDemo Android
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1439005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3993 4adac7df-926f-26a2-2b94-8c16560cd09d

Drop Virtual webcam check script as moved into buildbot scripts.

Having this script being located in the WebRTC repo doesn't make sense
since it has no connection to the source code.
Updating this script should apply to all build configurations and since
this script will be used for Chromium builders, we'll end up with having
to wait for a new WebRTC revision to be rolled in DEPS before it's updated.

TEST=none
BUG=none
TBR=phoglund

Review URL: https://webrtc-codereview.appspot.com/1444006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3992 4adac7df-926f-26a2-2b94-8c16560cd09d