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webrtc_m130/webrtc
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henrik.lundin@webrtc.org 245037df09 Remove default implementations for SuspendBelowMinBitrate 
These two methods had default implementations while waiting for
changes in libjingle to propagate. Now the changes are in, and
the default implementations are removed.

BUG=2436
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5222 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 12:01:45 +00:00
..
build
Remove the long disabled WEBRTC_SVNREVISION define.
2013-12-04 17:00:44 +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
Add callbacks for send channel rtcp statistics
2013-12-05 09:48:44 +00:00
system_wrappers
Remove ".." from include_dirs in build/common.
2013-11-11 10:20:27 +00:00
test
Adds support for sending redundant payloads over RTX.
2013-12-04 10:24:26 +00:00
tools
Made video quality toolchain more configurable.
2013-11-25 14:10:20 +00:00
video
Adds support for sending redundant payloads over RTX.
2013-12-04 10:24:26 +00:00
video_engine
Remove default implementations for SuspendBelowMinBitrate
2013-12-05 12:01:45 +00:00
voice_engine
Remove the long disabled WEBRTC_SVNREVISION define.
2013-12-04 17:00:44 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
call_tests.cc
Set local SSRC for VideoReceiveStream.
2013-12-03 10:13:04 +00:00
call.cc
Adds support for sending redundant payloads over RTX.
2013-12-04 10:24:26 +00:00
call.h
Adds support for sending redundant payloads over RTX.
2013-12-04 10:24:26 +00:00
common_types.h
Add callbacks for send channel rtcp statistics
2013-12-05 09:48:44 +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
experiments.h
Adds support for sending redundant payloads over RTX.
2013-12-04 10:24:26 +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
Set local SSRC for VideoReceiveStream.
2013-12-03 10:13:04 +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
Set local SSRC for VideoReceiveStream.
2013-12-03 10:13:04 +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
Adds support for sending redundant payloads over RTX.
2013-12-04 10:24:26 +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.