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pbos@webrtc.org e1fc3f22ea Disable check for all sent SSRCs being valid. 
Since the code for setting these up will set the codec before setting
SSRCs for the streams, any frames sent in between will be sent on
random-generated SSRCs.

This part should be added back during work on issue 1695.

BUG=1695
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5192 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-28 15:40:12 +00:00
..
build
Add -Wnon-virtual-dtor warning for C++ code.
2013-11-21 11:42:02 +00:00
common_audio
Upgrade scoped_ptr to Chromium's latest version.
2013-10-22 12:50:00 +00:00
common_video
Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420"
2013-10-31 23:41:04 +00:00
modules
This CL adds an API to enable robust validation of delay estimates.
2013-11-28 14:58:35 +00:00
system_wrappers
Remove ".." from include_dirs in build/common.
2013-11-11 10:20:27 +00:00
test
Set up SSRCs correctly after switching codec.
2013-11-28 11:59:31 +00:00
tools
Made video quality toolchain more configurable.
2013-11-25 14:10:20 +00:00
video
Disable check for all sent SSRCs being valid.
2013-11-28 15:40:12 +00:00
video_engine
Lock frame in ViECapturer::IncomingFrameI420.
2013-11-28 13:26:33 +00:00
voice_engine
Fixes a crash in VoE when unregistering JNI hooks.
2013-11-20 22:32:12 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
call_tests.cc
Implement and test EncodedImageCallback in new ViE API.
2013-11-26 11:41:59 +00:00
call.cc
Set up SSRCs correctly after switching codec.
2013-11-28 11:59:31 +00:00
call.h
Rename DestroyStream methods to include Video.
2013-11-21 13:49:43 +00:00
common_types.h
Make interface destructor virtual
2013-11-21 09:09:54 +00:00
common_unittest.cc
Fix unnamed-type-template-args warnings on clang.
2013-06-27 10:18:09 +00:00
common.h
Add a Config class interface to AudioProcessing for passing options.
2013-07-25 18:28:29 +00:00
config.cc
Rename RTP-extension constants.
2013-11-20 11:48:56 +00:00
config.h
Rename RTP-extension constants.
2013-11-20 11:48:56 +00:00
engine_configurations.h
Adds a new voice engine warning for the typing noise off state.
2013-09-17 18:09:20 +00:00
frame_callback.h
Implement and test EncodedImageCallback in new ViE API.
2013-11-26 11:41:59 +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
loopback.cc
Rename DestroyStream methods to include Video.
2013-11-21 13:49:43 +00:00
PATENTS
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
PRESUBMIT.py
Modified the presubmit checks such that difference license templates are checked for in webrtc and talk folder.
2013-07-22 22:32:50 +00:00
README.chromium
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Check if WARN_UNUSED_RESULT and COMPILE_ASSERT are defined.
2013-10-23 19:11:32 +00:00
video_engine_tests.isolate
Hook up audio/video sync to Call.
2013-11-18 11:45:11 +00:00
video_receive_stream.h
Rename video streams' start/stop methods.
2013-11-20 11:36:47 +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
Implement and test EncodedImageCallback in new ViE API.
2013-11-26 11:41:59 +00:00
webrtc_tests.gypi
Hook up audio/video sync to Call.
2013-11-18 11:45:11 +00:00
webrtc.gyp
Rename RTP-extension constants.
2013-11-20 11:48:56 +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.