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1272 Commits

Author SHA1 Message Date
pbos@webrtc.org
ae222b5be6 Remove dead code in WebRtcVideoEngine2 unittests. 
BUG=1788
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43609004

Cr-Commit-Position: refs/heads/master@{#8747}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8747 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-17 10:48:28 +00:00
magjed@webrtc.org
858024f1d9 WebRtcVideoFrame: Initialize members in empty constructor 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41319004

Cr-Commit-Position: refs/heads/master@{#8746}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8746 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-17 08:47:17 +00:00
pthatcher@webrtc.org
592470b4ff Remove a dependency of BaseChannel on WebRtcSession by having WebRtcSession push down new media descriptions to BaseChannel rather than having BaseChannel listen to the description changes from WebRtcSession. 
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47599004

Cr-Commit-Position: refs/heads/master@{#8743}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8743 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 21:16:23 +00:00
pthatcher@webrtc.org
6ad507ac35 Refactor how the TransportChannels are set in the BaseChannel to rely lesson Session, so that in the future we can rely on Transport instead, and also be able to change Transports on the fly for BUNDLE. 
Also, remove channel_name.  It's no longer needed.

This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/

R=decurtis@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43719004

Cr-Commit-Position: refs/heads/master@{#8741}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8741 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 20:19:42 +00:00
pthatcher@webrtc.org
4eeef584a7 Remove a hacky dependency of BaseChannel on BaseSession by moving the handling of DTLS setup failure into a signal on BaseChannel rather than a method call on BaseSession. 
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/

R=decurtis@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47589004

Cr-Commit-Position: refs/heads/master@{#8740}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8740 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 19:34:40 +00:00
pthatcher@webrtc.org
c04a97f054 Move from BaseSession::GetStats to WebRtcSession::GetTransportStats 
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/

Review URL: https://webrtc-codereview.appspot.com/45639004

Cr-Commit-Position: refs/heads/master@{#8739}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8739 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 19:32:23 +00:00
bjornv@webrtc.org
3f11823a1a Disables SW AEC when built-in AEC is enabled 
As of r7849 the built-in AEC on devicing supporting it is enabled by default.
Unfortunately, the SW AEC (AECM) was not disabled, hence running on top of the built-in one. This is not necessary. In fact it reduce double talk performance significantly.

BUG=4431
TESTED=manually
R=henrika@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49419004

Cr-Commit-Position: refs/heads/master@{#8735}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8735 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 14:22:17 +00:00
magjed@webrtc.org
2056ee3e3c Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*." 
This reverts commit r8731.

Reason for revert: Breakes Chromium FYI bots.

TBR=hbos, tommi

Review URL: https://webrtc-codereview.appspot.com/40359004

Cr-Commit-Position: refs/heads/master@{#8733}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8733 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 13:48:18 +00:00
hbos@webrtc.org
93d9d6503e I420VideoFrame.CreateFrame: Removed unnecessary buffer size arguments. 
R=magjed@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45629004

Cr-Commit-Position: refs/heads/master@{#8732}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8732 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 13:26:41 +00:00
hbos@webrtc.org
2dc5fa69b2 Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*. 
R=magjed@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40299004

Cr-Commit-Position: refs/heads/master@{#8731}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8731 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 13:02:19 +00:00
tommi@webrtc.org
4b89aa03bb Change StatsCollector to use DCHECK instead of ASSERT. 
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46579004

Cr-Commit-Position: refs/heads/master@{#8729}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8729 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 09:52:41 +00:00
kjellander@webrtc.org
eed2fcaa76 Roll chromium_revision 00e438c..8d51d96 (320241:320682) 
Relevant changes:
* src/third_party/android_tools: fd5a8ec..98a4345
Details: 00e438c..8d51d96/DEPS

This required updating our Android projects to API level 22,
as third_party/android_tools dropped support for API level 21.

Command used:
perl -pi -e "s/android-21/android-22/g" `find . -name project.properties`
Using 'android update project' would also work but that changes the
ANDROID_SDK_ROOT -> ANDROID_HOME, which the Chromium build toolchain
doesn't set properly when build/android/envsetup.sh is sourced.

Clang version was not updated in this roll.

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

Review URL: https://webrtc-codereview.appspot.com/42779004

Cr-Commit-Position: refs/heads/master@{#8728}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8728 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 09:00:41 +00:00
changbin.shao@webrtc.org
2d25b44f47 Check associated payload type when negotiate RTX codecs. 
At the moment, only payload name is checked when match two RTX codecs.
This will cause wrong behavior of codec negotiation if multiple RTX codecs
are added.

BUG=
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34189004

Cr-Commit-Position: refs/heads/master@{#8727}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8727 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 04:15:23 +00:00
tommi@webrtc.org
c29f7f3a5f Disable assert for nr of threads in PeerConnectionTest.java. 
This test is flaky so we need to figure out a better way to do it.
I've documented what we've observed and added a todo for myself to figure out a solution.

R=kjellander@webrtc.org
BUG=4424

Review URL: https://webrtc-codereview.appspot.com/46599004

Cr-Commit-Position: refs/heads/master@{#8725}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8725 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-14 18:15:47 +00:00
glaznev@webrtc.org
f1f558cde8 Fix AppRTCDemo and AppRTCDemoTest builds. 
On fresh checkout AppRTCDemo and corresponding tests
fail to build because resource file R.java is not auto generated properly.
On existing tree R.java will be picked up from previous
build leftover at talk/examples/android/gen.
Build bots did not detect this break for some reason.

R=wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43749004

Cr-Commit-Position: refs/heads/master@{#8723}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8723 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-14 02:48:47 +00:00
jiayl@webrtc.org
d83f4eff84 Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns. 
BUG=crbug/464995
R=pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8689

Committed: https://code.google.com/p/webrtc/source/detail?r=8701

Committed: https://code.google.com/p/webrtc/source/detail?r=8706

Review URL: https://webrtc-codereview.appspot.com/42659004

Cr-Commit-Position: refs/heads/master@{#8722}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8722 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-13 21:26:44 +00:00
pthatcher@webrtc.org
b01c707209 Use a NULL session in unit tests that don't actually use the session. 
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/

R=decurtis@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49379004

Cr-Commit-Position: refs/heads/master@{#8721}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8721 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-13 20:05:46 +00:00
pthatcher@webrtc.org
b4aac13810 Cleanup SocketMonitor a little so that it can handle a change in transport channel. And cleanup some names and style and such as well. 
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/

R=guoweis@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49399004

Cr-Commit-Position: refs/heads/master@{#8720}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8720 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-13 18:25:54 +00:00
pthatcher@webrtc.org
990a00c30a Remove unused transport code. 
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49389004

Cr-Commit-Position: refs/heads/master@{#8719}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8719 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-13 18:20:48 +00:00
minyue@webrtc.org
9b2e1144df Supporting Opus DTX in Voice Engine. 
Opus DTX is an Opus specific feature. It does not require WebRTC VAD/DTX, therefore is not set by VoECodec::SetVADStatus(), but rather a dedicated API.

BUG=1014
R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43709004

Cr-Commit-Position: refs/heads/master@{#8716}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8716 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-13 09:38:55 +00:00
kjellander@webrtc.org
503a9e822a Make AppRTCDemoTest pass without Internet connection. 
The AppRTCDemoTest is failing if the Android device lacks
an Internet connection (e.g. is in flight mode).
This change makes it benefit from the work done in
https://review.webrtc.org/36769004/ to work around that
limitation for loopback tests.

R=phoglund@webrtc.org
TBR=glaznev@webrtc.org
BUG=4421
TESTED=Successful run on Nexus 7 (2013) in flight mode using:
ninja -C out/Release
. build/android/envsetup.sh
adb install -r out/Release/apks/AppRTCDemo.apk
webrtc/build/android/test_runner.py instrumentation --test-apk AppRTCDemoTest --verbose --release

Review URL: https://webrtc-codereview.appspot.com/45649004

Cr-Commit-Position: refs/heads/master@{#8714}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8714 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-13 09:06:58 +00:00
jiayl@webrtc.org
8372888b07 Revert "Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns." 
This reverts commit 45bc01a7172402aa4bb8d457474300533c273413.
TBR=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/47559004

Cr-Commit-Position: refs/heads/master@{#8711}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8711 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-13 00:32:43 +00:00
glaznev@webrtc.org
3d3c005f36 Fix Android peer connection client instrumentation tests. 
- Updated Java VideoRenderer removes setSize() from video renderer interface.
Remove no longer valid test, which requires setSize() call before any
frame can be rendered.
- test_runner.py tries to run private member of InstrumentationTestCase class.
Workaround it by renaming private loopback test method.

R=wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47549004

Cr-Commit-Position: refs/heads/master@{#8707}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8707 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-12 23:07:17 +00:00
jiayl@webrtc.org
fde1de93f9 Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns. 
BUG=crbug/464995
R=pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8689

Committed: https://code.google.com/p/webrtc/source/detail?r=8701

Review URL: https://webrtc-codereview.appspot.com/42659004

Cr-Commit-Position: refs/heads/master@{#8706}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8706 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-12 23:02:13 +00:00
guoweis@webrtc.org
00c509ad1c Add concept of whether video renderer supports rotation. 
Rotation is best done when rendered in GPU, added the shader code which rotates the frame. For renderers which don't support rotation, the rotation will be done before sending down the frame to render. By default, assume renderer can't do rotation.

Tested with peerconnection_client on windows, AppRTCDemo on Mac.

BUG=4145
R=glaznev@webrtc.org, pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8660

Committed: https://code.google.com/p/webrtc/source/detail?r=8661

Review URL: https://webrtc-codereview.appspot.com/43569004

Cr-Commit-Position: refs/heads/master@{#8705}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8705 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-12 21:38:19 +00:00
jiayl@webrtc.org
04cd69887d Revert "Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns." 
This reverts commit 93604daf0e.

TBR=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/40329004

Cr-Commit-Position: refs/heads/master@{#8704}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8704 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-12 21:36:42 +00:00
guoweis@webrtc.org
fdd1057949 Add CVO support to Vie layer. 
1. standard plumbing CVO through vie layer.
2. added a rtp_cvo.h which has both conversion functions from rtp header byte to/from VideoRotation.

WebRTCVideoEngine will later pass the rotation info in SendFrame() through VieVideoFrameI420.

BUG=4145
R=mflodman@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46429007

Cr-Commit-Position: refs/heads/master@{#8703}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8703 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-12 20:51:50 +00:00
guoweis@webrtc.org
4f85288e71 Socket options are only applied when first setting TransportChannelImpl. 
Also fixed the issue when we have an TransportChannelImpl, the socket
option is not preserved.

Since this is a code path that will be modified by bundle (which Peter also has a test case already), we don't need a test case here.

BUG=4374
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42699004

Cr-Commit-Position: refs/heads/master@{#8702}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8702 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-12 20:10:22 +00:00
jiayl@webrtc.org
93604daf0e Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns. 
BUG=crbug/464995
R=pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8689

Review URL: https://webrtc-codereview.appspot.com/42659004

Cr-Commit-Position: refs/heads/master@{#8701}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8701 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-12 19:05:22 +00:00
tommi@webrtc.org
d3900296ae Use a variant for storing stats values in StatsCollector code. 
This cuts down on the amount of string copying we currently do and paves the way for separating the code that fetches the stats from the code that populates the stats reports.  As is, that code is intertwined, so we populate the stats on both signaling and worker thread.

I'm also adding some documentation and TODOs for further improvements.

BUG=2822
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47459004

Cr-Commit-Position: refs/heads/master@{#8700}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8700 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-12 16:36:15 +00:00
tommi@webrtc.org
75b7f17c29 Temporary interface change to StatsReport::Id. 
This change is just to allow rolling into Chromium, update Chromium and then commit the actual change in WebRTC that requires the interface change. It allows using a StatsReport::Id object as a pointer (foo->Bar()), since in an upcoming change, Id objects will be pointers.

R=magjed@webrtc.org
BUG=2822

Review URL: https://webrtc-codereview.appspot.com/43689004

Cr-Commit-Position: refs/heads/master@{#8697}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8697 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-12 13:31:24 +00:00
magjed@webrtc.org
afdd5dd372 Revert "Revert "Remove frame copy from cricket::VideoFrame to I420VideoFrame"" 
This reverts r8683 and is a reland of r8682.

Reason for revert: The thread checker in Chromium that crashed has been fixed now.

BUG=1128
TBR=tommi,pbos,pthatcher

Review URL: https://webrtc-codereview.appspot.com/40319004

Cr-Commit-Position: refs/heads/master@{#8696}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8696 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-12 13:11:43 +00:00
mflodman@webrtc.org
e9413c686e Revert 8689 "Fix an issue in DtlsIdentityStore when the store is..." 
> Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns.
> 
> BUG=crbug/464995
> R=pthatcher@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/42659004

TBR=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42729004

Cr-Commit-Position: refs/heads/master@{#8690}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8690 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-12 08:42:18 +00:00
jiayl@webrtc.org
2a3942adc6 Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns. 
BUG=crbug/464995
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42659004

Cr-Commit-Position: refs/heads/master@{#8689}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8689 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-11 23:33:42 +00:00
decurtis@webrtc.org
8c5ea8a811 Fix temporal layer log string. 
BUG=
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43639004

Patch from Noah Richards <noahric@chromium.org>.

Cr-Commit-Position: refs/heads/master@{#8687}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8687 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-11 19:59:43 +00:00
glaznev@webrtc.org
ae1a078ac4 Convert AppRTCDemo and AppRTCDemoTest to proper GYP target. 
Initial CL for converting AppRTCDemo and AppRTCDemoTest to
the Chromium style of APK targets. This would
make it possible to get rid of all the ugly
bash stuff we currently have.

CL will bump minimum SDK to v14, but this is the requirement to use Chrome tools.

Initial work was done by kjellander@
https://webrtc-codereview.appspot.com/44549005/

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43679004

Cr-Commit-Position: refs/heads/master@{#8686}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8686 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-11 17:52:44 +00:00
magjed@webrtc.org
b218ff5531 Revert "Remove frame copy from cricket::VideoFrame to I420VideoFrame" 
This reverts r8682.

Reason for revert: Fails on Chromium FYI content_browsertests

BUG=1128
TBR=tommi,pbos,pthatcher

Review URL: https://webrtc-codereview.appspot.com/47529004

Cr-Commit-Position: refs/heads/master@{#8683}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8683 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-11 15:29:59 +00:00
magjed@webrtc.org
370a72cc3f Remove frame copy from cricket::VideoFrame to I420VideoFrame 
BUG=1128
R=pbos@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42249004

Cr-Commit-Position: refs/heads/master@{#8682}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8682 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-11 14:16:16 +00:00
pbos@webrtc.org
e77c9c8df5 Build WebRtcMediaEngine2 outside of Chromium. 
Removes #ifdef WEBRTC_CHROMIUM_BUILD from
talk/media/webrtc/webrtcmediaengine.cc. WebRtcVideoEngine2 is built on
all platforms so there's no longer any need to guard this code under
ifdefs.

BUG=1788
R=sprang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42719004

Cr-Commit-Position: refs/heads/master@{#8679}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8679 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-11 10:50:16 +00:00
braveyao@webrtc.org
9bfa5f0405 In r8605, DTLS is enabled by default for native webrtc. So we have to disable it explicitly in peerconnection example for loopback test. 
BUG=4386
TEST=Manual Test
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44599004

Cr-Commit-Position: refs/heads/master@{#8677}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8677 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-11 03:21:41 +00:00
glaznev@webrtc.org
fc516077ed Fix Android AppRTCDemo failure on devices with one or no camera. 
- Disable video call on devices with no camera.
- Open default camera and disable camera switch on
devices with one camera.

BUG=4373
R=braveyao@webrtc.org, wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46539004

Cr-Commit-Position: refs/heads/master@{#8674}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8674 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-10 18:21:19 +00:00
magjed@webrtc.org
4052d88162 Remove GetLastRenderedFrame 
This function is not used.

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

Review URL: https://webrtc-codereview.appspot.com/40269004

Cr-Commit-Position: refs/heads/master@{#8673}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8673 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-10 16:36:42 +00:00
magjed@webrtc.org
d7452a0168 Revert "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame." 
This reverts commit r8633.

Reason for revert: Performance regressions in browser_tests_new_vie and webrtc_perf_tests.

BUG=1128,chromium:465287,chromium:465306
TBR=pbos,mflodman,perkj

Review URL: https://webrtc-codereview.appspot.com/46549004

Cr-Commit-Position: refs/heads/master@{#8670}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8670 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-10 15:13:13 +00:00
hbos@webrtc.org
aa57702c08 Removed texture_video_frame.h and webrtctexturevideoframe.h 
BUG=1128
R=magjed@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45579004

Cr-Commit-Position: refs/heads/master@{#8667}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8667 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-10 09:04:18 +00:00
guoweis@webrtc.org
f9a75d99b9 Revert "Add concept of whether video renderer supports rotation." 
This reverts commit 0ad48935fc.

TBR=guoweis@webrtc.org

BUG=

Review URL: https://webrtc-codereview.appspot.com/41199004

Cr-Commit-Position: refs/heads/master@{#8663}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8663 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-10 06:37:41 +00:00
guoweis@webrtc.org
60a2aa0652 Revert "Add concept of whether video renderer supports rotation." 
This reverts commit 31d16467ac.

TBR=guoweis@webrtc.org

BUG=

Review URL: https://webrtc-codereview.appspot.com/47489004

Cr-Commit-Position: refs/heads/master@{#8662}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8662 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-10 06:20:18 +00:00
guoweis@webrtc.org
31d16467ac Add concept of whether video renderer supports rotation. 
Rotation is best done when rendered in GPU, added the shader code which rotates the frame. For renderers which don't support rotation, the rotation will be done before sending down the frame to render. By default, assume renderer can't do rotation.

BUG=4145
R=glaznev@webrtc.org, pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8660

Review URL: https://webrtc-codereview.appspot.com/43569004

Cr-Commit-Position: refs/heads/master@{#8661}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8661 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-10 06:11:44 +00:00
guoweis@webrtc.org
0ad48935fc Add concept of whether video renderer supports rotation. 
Rotation is best done when rendered in GPU, added the shader code which rotates the frame. For renderers which don't support rotation, the rotation will be done before sending down the frame to render. By default, assume renderer can't do rotation.

BUG=4145
R=glaznev@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43569004

Cr-Commit-Position: refs/heads/master@{#8660}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8660 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-10 02:43:40 +00:00
kwiberg@webrtc.org
67186fe00c Fix clang style warnings in webrtc/base 
Mostly this consists of marking functions with override when
applicable, and moving function bodies from .h to .cc files.

Not inlining virtual functions with simple bodies such as

  { return false; }

strikes me as probably losing more in readability than we gain in
binary size and compilation time, but I guess it's just like any other
case where enabling a generally good warning forces us to write
slightly worse code in a couple of places.

BUG=163
R=kjellander@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47429004

Cr-Commit-Position: refs/heads/master@{#8656}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8656 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-09 22:24:25 +00:00
glaznev@webrtc.org
2989204130 Fix instability in peer connection client unit test. 
- Add a separate thread to process peer connection ICE messages
to void setting remote ICe candidate in local ICE candidate callback.
- Set proper constraints values.

R=wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42279004

Cr-Commit-Position: refs/heads/master@{#8655}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8655 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-09 19:15:32 +00:00