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d6a0efdc86f61b198b800dba301f53f847eb9ba0
platform-external-webrtc/webrtc
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fischman@webrtc.org d6a0efdc86 VideoCaptureAndroid: quit & join the camera thread on stopCapture. 
Also fix latent bug where setPreviewRotation() wouldn't hold
the lock while its delegate setPreviewRotationOnCameraThread()
was running, allowing the camera to be freed between the
null-check and the use.

BUG=3389
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17619007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6266 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-28 18:37:07 +00:00
..
base
Rebase webrtc/base with r6232:
2014-05-23 18:40:46 +00:00
build
removed webrtc_base_tests_utils from merge libs as it was breaking some builds.
2014-05-15 21:45:09 +00:00
common_audio
common_audio/signal_processing: Fixes arm compilation issues with gcc 4.8
2014-05-28 08:45:04 +00:00
common_video
Add support of texture frames for video capturer.
2014-05-28 07:00:51 +00:00
examples
WebRTCDemo: clean the error message due to API clean up and add ability to route the audio through all three outputs, headset/earpiece/loudspeaker
2014-05-21 03:37:45 +00:00
modules
VideoCaptureAndroid: quit & join the camera thread on stopCapture.
2014-05-28 18:37:07 +00:00
overrides/webrtc/base
Rename webrtc/base's IS_ALIGNED macro to RTC_IS_ALIGNED to avoid conflict between webrtc/base/basictypes.h and third_party/.../vpx_codec.h.
2014-05-21 16:52:14 +00:00
system_wrappers
Adding R/W lock to SimulatedClock
2014-05-27 09:34:58 +00:00
test
Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
2014-05-21 21:18:46 +00:00
tools
Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
2014-05-21 21:18:46 +00:00
video
First incoming packet was not accounted for in receive stats. Changed call order for incoming packet to receive statistics class.
2014-05-26 13:06:04 +00:00
video_engine
Add support of texture frames for video capturer.
2014-05-28 07:00:51 +00:00
voice_engine
This CL is to adding feedback of packet loss rate to encoder in voice engine. A direct reason for doing it is to make use of Opus FEC, which can adapt itself to changes in the packet loss rate.
2014-05-28 09:52:06 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
call.h
Add DeliveryStatus enum to DeliverPacket().
2014-05-14 13:57:12 +00:00
common_types.h
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00
common.gyp
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00
common.h
Add a Config class interface to AudioProcessing for passing options.
2013-07-25 18:28:29 +00:00
config.cc
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00
config.h
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00
engine_configurations.h
Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs.
2014-05-12 12:19:19 +00:00
experiments.h
Adding API for setting bandwidth estimation configurations.
2014-03-25 10:37:31 +00:00
frame_callback.h
Wire up statistics in video receive stream of new API
2014-02-07 12:06:29 +00:00
LICENSE
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY
Consolidate all third party licenses in LICENSE_THIRD_PARTY.
2013-05-05 18:54:10 +00:00
OWNERS
Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
2014-04-14 20:08:03 +00:00
PATENTS
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
PRESUBMIT.py
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
README.chromium
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
supplement.gypi
Roll chromium_revision 260462:266514
2014-04-29 09:36:40 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Remove ALLOW_UNUSED.
2014-05-05 18:18:02 +00:00
video_engine_tests.isolate
Merge metrics_unittests into video_engine_tests.
2013-12-13 14:31:47 +00:00
video_receive_stream.h
Rename Start/Stop in Video{Send,Receive}Streams.
2014-04-24 11:13:21 +00:00
video_renderer.h
Separate Call API/build files from video_engine/.
2013-10-28 16:32:01 +00:00
video_send_stream.h
Revert "Revert "Remove VideoSendStreamInput::PutFrame.""
2014-05-23 13:03:45 +00:00
webrtc_examples.gyp
Add webrtc field trials API.
2014-05-14 12:24:04 +00:00
webrtc_perf_tests.isolate
Move realtime tests to webrtc_perf_tests.
2013-12-13 12:48:05 +00:00
webrtc_tests.gypi
Revert 6257 "Rename neteq4 folder to neteq"
2014-05-28 07:33:39 +00:00
webrtc.gyp
Add ToString() to VideoSendStream::Config.
2014-05-15 09:35:06 +00:00

README.chromium 

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.