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fischman@webrtc.org f4e070eca5 Added auto-call feature to WebRTCDemo. 
This (compile-time switchable) option automatically starts & stops calls in
series to stress-test the setup/teardown codepaths.  When startCPULoad() is
removed (https://webrtc-codereview.appspot.com/972008/) this showed no
hangs/crashes after completing 200 start/stop pairs.

Also fixed a tiny shutdown-order bug (onDestroy() calling super.onDestroy()
before performing self-shutdown) and changed default video frame resolution to
640x480 to more effectively stress the device (and be a more compelling demo).

BUG=1162

Review URL: https://webrtc-codereview.appspot.com/939032

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3238 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-04 16:53:43 +00:00
..
build
Enable java soundcard impl as the default
2012-11-29 03:13:00 +00:00
common_audio
Porting ARM optimization from Android to ios.
2012-11-17 00:22:46 +00:00
common_video
Fixed standard PSNR/SSIM test.
2012-11-29 10:08:16 +00:00
modules
Dual-stream implementation, not including VoE APIs.
2012-12-03 22:13:31 +00:00
system_wrappers
Reformatted data_log.
2012-12-03 09:01:21 +00:00
test
Removed stale version of fuzzer; it's now internal.
2012-12-03 14:28:40 +00:00
tools
Fixed bug that caused frame_cutter_unittest to fail when built with MVS2008.
2012-11-30 12:37:14 +00:00
video_engine
Added auto-call feature to WebRTCDemo.
2012-12-04 16:53:43 +00:00
voice_engine
Revert 3231 - VoE Changes to enable dual_streaming.
2012-12-04 10:02:02 +00:00
common_types.h
Add a kTraceTerseInfo level for non-verbose logging.
2012-11-20 07:34:45 +00:00
engine_configurations.h
First pass of MediaCodecDecoder which uses Android MediaCodec API.
2012-12-04 06:38:19 +00:00
LICENSE
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
PATENTS
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
README.chromium
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
typedefs.h
Move src/ -> webrtc/
2012-10-22 18:19:23 +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.