Commit Graph

97 Commits

Author SHA1 Message Date
tommi@webrtc.org
5c3ee4bce6 Add empty implementation file that will hold statstypes.h implementation.
The implementation for the types currently in statstypes.h is split between statstypes.h and statscollector.cc.

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7844 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 10:47:01 +00:00
andrew@webrtc.org
3b3c406908 Revert 7826 "Change Android PeerConnectionUnittest to build usin..."
Broke gclient runhooks on internal bots. e.g.
http://chromegw/i/internal.client.webrtc/builders/Linux64%20Debug/builds/3575

> Change Android PeerConnectionUnittest to build using Chrome macros.
> The purpose is to be able to run the tests using Chromes buildbots. To run:
> CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py instrumentation --test-apk=libjingle_peerconnection_android_unittest
> 
> This also add a new build target to build java PeerConnection using Chromes build macros.
> 
> BUG=4031
> R=kjellander@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/28189004

TBR=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7829 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 17:21:50 +00:00
perkj@webrtc.org
ed7824b1c0 Change Android PeerConnectionUnittest to build using Chrome macros.
The purpose is to be able to run the tests using Chromes buildbots. To run:
CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py instrumentation --test-apk=libjingle_peerconnection_android_unittest

This also add a new build target to build java PeerConnection using Chromes build macros.

BUG=4031
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7826 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 15:41:01 +00:00
magjed@webrtc.org
f58b455cf7 cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after.
In VideoCapturer::OnFrameCaptured, we currently convert cricket::CapturedFrame to cricket::VideoFrame and then send that to VideoAdapter::AdaptFrame. AdaptFrame may then decide to drop the frame. It would be faster to drop the frame before converting to cricket::VideoFrame.

This CL refactors VideoAdapter with a new function AdaptFrameResolution that takes captured resolution as input and output adapted resolution, or 0x0 if the frame should be dropped. Using that function, frames can be dropped before any conversion takes place.

R=fbarchard@google.com, perkj@webrtc.org, tommi@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7702

Committed: https://code.google.com/p/webrtc/source/detail?r=7707

Review URL: https://webrtc-codereview.appspot.com/29949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7721 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-19 18:09:14 +00:00
tommi@webrtc.org
4ec19e306a Revert 7707 "cricket::VideoAdapter: Drop frames before spending ..."
This didn't compile on the FYI bots.  Example error:

FAILED: E:\b\depot_tools\python276_bin\python.exe gyp-win-tool link-with-manifests environment.x86 True chrome_child.dll "E:\b\depot_tools\python276_bin\python.exe gyp-win-tool link-wrapper environment.x86 False link.exe /nologo /IMPLIB:chrome_child.dll.lib /DLL /OUT:chrome_child.dll @chrome_child.dll.rsp" 2 mt.exe rc.exe "obj\chrome\chrome_child_dll.chrome_child.dll.intermediate.manifest" obj\chrome\chrome_child_dll.chrome_child.dll.generated.manifest
content_renderer.lib(content_renderer.webrtc_video_capturer_adapter.obj) : error LNK2001: unresolved external symbol "public: virtual class cricket::VideoFrame * __thiscall cricket::VideoFrameFactory::CreateAliasedFrame(struct cricket::CapturedFrame const *,int,int,int,int)const " (?CreateAliasedFrame@VideoFrameFactory@cricket@@UBEPAVVideoFrame@2@PBUCapturedFrame@2@HHHH@Z)

libjingle_webrtc_common.lib(libjingle_webrtc_common.peerconnectionfactory.obj) : error LNK2001: unresolved external symbol "public: virtual class cricket::VideoFrame * __thiscall cricket::VideoFrameFactory::CreateAliasedFrame(struct cricket::CapturedFrame const *,int,int,int,int)const " (?CreateAliasedFrame@VideoFrameFactory@cricket@@UBEPAVVideoFrame@2@PBUCapturedFrame@2@HHHH@Z)

libjingle_webrtc_common.lib(libjingle_webrtc_common.videocapturer.obj) : error LNK2001: unresolved external symbol "public: virtual class cricket::VideoFrame * __thiscall cricket::VideoFrameFactory::CreateAliasedFrame(struct cricket::CapturedFrame const *,int,int,int,int)const " (?CreateAliasedFrame@VideoFrameFactory@cricket@@UBEPAVVideoFrame@2@PBUCapturedFrame@2@HHHH@Z)

libjingle_webrtc_common.lib(libjingle_webrtc_common.dummydevicemanager.obj) : error LNK2001: unresolved external symbol "public: virtual class cricket::VideoFrame * __thiscall cricket::VideoFrameFactory::CreateAliasedFrame(struct cricket::CapturedFrame const *,int,int,int,int)const " (?CreateAliasedFrame@VideoFrameFactory@cricket@@UBEPAVVideoFrame@2@PBUCapturedFrame@2@HHHH@Z)

chrome_child.dll : fatal error LNK1120: 1 unresolved externals


> cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after.
> 
> In VideoCapturer::OnFrameCaptured, we currently convert cricket::CapturedFrame to cricket::VideoFrame and then send that to VideoAdapter::AdaptFrame. AdaptFrame may then decide to drop the frame. It would be faster to drop the frame before converting to cricket::VideoFrame.
> 
> This CL refactors VideoAdapter with a new function AdaptFrameResolution that takes captured resolution as input and output adapted resolution, or 0x0 if the frame should be dropped. Using that function, frames can be dropped before any conversion takes place.
> 
> R=fbarchard@google.com, perkj@webrtc.org, tommi@webrtc.org
> 
> Committed: https://code.google.com/p/webrtc/source/detail?r=7702
> 
> Review URL: https://webrtc-codereview.appspot.com/29949004

TBR=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7708 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-16 22:58:11 +00:00
magjed@webrtc.org
858dbbced2 cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after.
In VideoCapturer::OnFrameCaptured, we currently convert cricket::CapturedFrame to cricket::VideoFrame and then send that to VideoAdapter::AdaptFrame. AdaptFrame may then decide to drop the frame. It would be faster to drop the frame before converting to cricket::VideoFrame.

This CL refactors VideoAdapter with a new function AdaptFrameResolution that takes captured resolution as input and output adapted resolution, or 0x0 if the frame should be dropped. Using that function, frames can be dropped before any conversion takes place.

R=fbarchard@google.com, perkj@webrtc.org, tommi@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7702

Review URL: https://webrtc-codereview.appspot.com/29949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7707 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-16 18:21:51 +00:00
magjed@webrtc.org
a73d746562 Revert 7702 "cricket::VideoAdapter: Drop frames before spending ..."
Rease for revert: failed internal test cases

> cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after.
> 
> In VideoCapturer::OnFrameCaptured, we currently convert cricket::CapturedFrame to cricket::VideoFrame and then send that to VideoAdapter::AdaptFrame. AdaptFrame may then decide to drop the frame. It would be faster to drop the frame before converting to cricket::VideoFrame.
> 
> This CL refactors VideoAdapter with a new function AdaptFrameResolution that takes captured resolution as input and output adapted resolution, or 0x0 if the frame should be dropped. Using that function, frames can be dropped before any conversion takes place.
> 
> R=fbarchard@google.com, perkj@webrtc.org, tommi@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/29949004

TBR=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7703 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-14 13:25:25 +00:00
magjed@webrtc.org
bbd8cad21f cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after.
In VideoCapturer::OnFrameCaptured, we currently convert cricket::CapturedFrame to cricket::VideoFrame and then send that to VideoAdapter::AdaptFrame. AdaptFrame may then decide to drop the frame. It would be faster to drop the frame before converting to cricket::VideoFrame.

This CL refactors VideoAdapter with a new function AdaptFrameResolution that takes captured resolution as input and output adapted resolution, or 0x0 if the frame should be dropped. Using that function, frames can be dropped before any conversion takes place.

R=fbarchard@google.com, perkj@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7702 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-14 12:10:46 +00:00
tkchin@webrtc.org
8125744a5f Cleanup RTCVideoRenderer interface.
RTCVideoRenderer should be a protocol not a class. This change includes
an adapter for use with the C++ apis. The video views have been refactored
to implement that protocol.

BUG=3795
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7626 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 23:06:15 +00:00
henrike@webrtc.org
269fb4bc90 move xmpp and p2p to webrtc
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and
webrtc/p2p. Also makes libjingle use those version instead of the one in the talk folder.

BUG=3379

Review URL: https://webrtc-codereview.appspot.com/26999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7549 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 22:20:11 +00:00
asapersson@webrtc.org
580d367b14 Add macros and APIs for webrtc histograms.
BUG=crbug/419657

Code that links system_wrappers.gyp:system_wrappers should either:
- provide implementations for the APIs, or
- link with default implementations in system_wrappers.gyp:system_wrappers_default.

R=andresp@webrtc.org, kjellander@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7508 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 12:57:56 +00:00
henrike@webrtc.org
28100cb388 Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p."
BUG=N/A
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7472 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-17 22:03:39 +00:00
henrike@webrtc.org
d1ba6d9cbf Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p.
BUG=3379
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27709005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7459 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-15 17:30:28 +00:00
andresp@webrtc.org
ab071daab8 Split video_render_module implementation into default and internal implementation.
Targets must now link with implementation of their choice instead of at "gyp"-time.

Targets linking with libjingle_media:
- internal implementation when build_with_chromium=0, default otherwise.

Targets linking with default render implementation:
- video_engine_tests
- video_loopback
- video_replay
- anything dependent on webrtc_test_common

Targets linking with internal render implementation:
- vie_auto_test
- video_render_tests
- libwebrtcdemo-jni
- video_engine_core_unittests

GN changes:
- Not many since there is almost no test definitions.

Work-around for chromium:
- Until chromium has updated libpeerconnection to link with video_capture_impl and video_render_impl, webrtc target automatically depends on it. This should fix the FYI bots and not require a webrtc roll to fix.

Re-enable android tests by reverting 7026 (some tests left disabled).

TESTED: passes all the bots. If this inadvertently breaks a target please fix the linking rules so the target has the desired implementation linked in.
BUG=3770
R=kjellander@webrtc.org, pbos@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7217 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-18 08:58:15 +00:00
andresp@webrtc.org
a74eda1b6f Split video_capture_module specific implementation (external vs internal capture)
into its own targets. Dependencies must link directly with the desired one.

Targets linking with libjingle_media:
 - internal implementation when build_with_chromium=0, default otherwise.

Targets linking with default/external capture implementation:
 - anything dependent on webrtc_test_common
 - anything dependent on video_engine_core

Targets linking with internal capture implementation:
 - vie_auto_test
 - anything dependent on webrtc_test_renderer

GN changes:
 - Not many since there is almost no test definitions.

TESTED: passes all the bots. If this inadvertently breaks a target please fix the linking rules so the target has the desired implementation linked in.

BUG=3768
R=glaznev@webrtc.org
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7209 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17 11:50:19 +00:00
henrike@webrtc.org
7f826350e3 Stop building talk/xmllite since it is no longer used.
BUG=3379
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7176 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-15 08:13:36 +00:00
buildbot@webrtc.org
a42a3ade54 (Auto)update libjingle 75390072-> 75428737
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7174 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-13 01:09:18 +00:00
henrike@webrtc.org
1d8f780779 Stop building talk/sound since it is no longer used.
BUG=N/A
R=pbos@webrtc.org
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7156 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 17:16:56 +00:00
sprang@webrtc.org
c665dcb205 Revert 7145 "Stop building talk/sound since it is no longer used."
> Stop building talk/sound since it is no longer used.
> 
> BUG=N/A
> R=pbos@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/22319004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7148 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 08:29:53 +00:00
henrike@webrtc.org
4c876453c8 Stop building talk/sound since it is no longer used.
BUG=N/A
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7145 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10 22:18:04 +00:00
henrike@webrtc.org
b2efb6771c Put base tests in webrtc_tests.gyp
BUG=N/A
R=andrew@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7140 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10 17:28:19 +00:00
pbos@webrtc.org
bcb6bcfe6c Remove HybridVideoEngine.
This is currently unused dead code.

R=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/24409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7055 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 07:32:26 +00:00
buildbot@webrtc.org
fa4535b270 (Auto)update libjingle 74694022-> 74696326
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7045 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 16:49:04 +00:00
buildbot@webrtc.org
573a1eef3d (Auto)update libjingle 74202294-> 74230205
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6990 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-27 17:21:19 +00:00
henrike@webrtc.org
0481f15f02 (Auto)update libjingle 73399579-> 73626167
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6928 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-19 14:56:59 +00:00
buildbot@webrtc.org
65b98d12c3 (Auto)update libjingle 72839629-> 72847605
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6854 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-07 22:09:08 +00:00
buildbot@webrtc.org
5b1ebacca2 (Auto)update libjingle 72820109-> 72822008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6850 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-07 17:18:00 +00:00
buildbot@webrtc.org
d509678a4e (Auto)update libjingle 72819313-> 72820109
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6849 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-07 16:57:07 +00:00
buildbot@webrtc.org
94b996cc18 (Auto)update libjingle 72785516-> 72819313
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6848 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-07 16:47:14 +00:00
buildbot@webrtc.org
476efa2031 (Auto)update libjingle 72785180-> 72785516
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6842 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-07 04:55:21 +00:00
buildbot@webrtc.org
4f0d401fae (Auto)update libjingle 72682155-> 72785180
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6841 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-07 04:47:36 +00:00
buildbot@webrtc.org
8e885990ae (Auto)update libjingle 72566057-> 72591796
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6824 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-04 23:56:14 +00:00
buildbot@webrtc.org
e0d03f13e4 (Auto)update libjingle 72443101-> 72446860
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6815 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-02 03:04:01 +00:00
buildbot@webrtc.org
6e203d50a3 (Auto)update libjingle 72442050-> 72443101
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6814 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-02 01:13:04 +00:00
buildbot@webrtc.org
52148c2f74 (Auto)update libjingle 72430895-> 72442050
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6813 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-02 00:56:56 +00:00
buildbot@webrtc.org
7cb60ccae1 (Auto)update libjingle 72407428-> 72430895
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6812 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-01 22:03:36 +00:00
buildbot@webrtc.org
3bc48247b7 (Auto)update libjingle 72403605-> 72407428
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6811 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-01 16:53:32 +00:00
henrike@webrtc.org
d9843da9ee libjingle: stop building files in talk/base as they are no longer used as of r6799
BUG=3379
R=thorcarpenter@google.com

Review URL: https://webrtc-codereview.appspot.com/16189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6802 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-30 04:00:52 +00:00
buildbot@webrtc.org
d4e598d57a (Auto)update libjingle 72097588-> 72159069
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-29 17:36:52 +00:00
glaznev@webrtc.org
efe4b9af49 Add VP8 video decoding hw acceleration support to Java Peerconnection library.
For now NVidia decoder is supported only,
Qualcomm will be added once b/16353967 is fixed.

TODO:
- Support queuing 2-3 decoder input buffers.
- Add average decoding time statistics.
- Add Qualcomm hw decoder support.

BUG=3030
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6758 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-22 17:44:53 +00:00
tkchin@webrtc.org
b038c72369 Enable SCTP compile for iOS.
Chromium's been updated to pull a version of usrsctplib that will compile correctly. This update DEPS to point at new revision and turn on the compile time flags for iOS sctp.

BUG=3211
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6681 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-14 20:24:09 +00:00
buildbot@webrtc.org
bb2d65895b (Auto)update libjingle 69617317-> 69623266
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6508 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-20 14:58:56 +00:00
buildbot@webrtc.org
58e7c8660c (Auto)update libjingle 69588980-> 69589535
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6503 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-20 00:26:50 +00:00
kjellander@webrtc.org
6b6e58d632 Remove unused test_env.py from isolate files + fix nss path.
This is not necessary for executing tests for WebRTC.
It probably appeared in our .isolate files because of code
copied from Chromium.

BUG=
TEST=All non-baremetal trybots passing.
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6373 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-09 14:35:09 +00:00
glaznev@webrtc.org
c3288c130d Add OpenGL Android video renderer which can display multiple
yuv420 images in a single GLSurfaceView.
Start using new video renderer in AppRTC demo app.

BUG=
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6360 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-06 21:57:46 +00:00
pbos@webrtc.org
289a35c56d Add empty webrtcmediaengine.cc.
Should contain CreateWebRtcMediaEngine as soon as
libjingle/libjingle.gyp in Chromium builds this file. This file is added
ahead of time to get a smoother rolling process.

BUG=1788
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19599005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6313 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-03 14:51:34 +00:00
tkchin@webrtc.org
acca675bcf Implement mac version of AppRTCDemo.
- Refactored and moved AppRTCDemo to support sharing AppRTC connection code between iOS and mac counterparts.
- Refactored OpenGL rendering code to be shared between iOS and mac counterparts.
- iOS AppRTCDemo now respects video aspect ratio.

BUG=2168
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6291 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-30 22:26:06 +00:00
tkchin@webrtc.org
1732a591e7 Add a UIView for rendering a video track.
RTCEAGLVideoView provides functionality to render a supplied RTCVideoTrack using OpenGLES2.

R=fischman@webrtc.org
BUG=3188

Review URL: https://webrtc-codereview.appspot.com/12489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6192 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-19 23:26:01 +00:00
andresp@webrtc.org
581e2172af Fix libjingle to provide a field_trial implementation.
This completes https://webrtc-codereview.appspot.com/14489004/ by updating libjingle rules.

BUG=crbug/367114
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6149 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 13:12:45 +00:00
pbos@webrtc.org
b5a22b1464 Revert r6110 and r6109.
Effectively re-landing r6104 as well as r6108 which includes a fix to a
compile error that caused r6104 to be reverted in r6110.

BUG=
TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6119 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 11:07:01 +00:00