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asapersson@webrtc.org 86bb56a7f5 Revert 5228 "Use the RTT from RtcpRttStats class if provided whe..." 
> Use the RTT from RtcpRttStats class if provided when sending/receiving NACK.
> 
> R=holmer@google.com
> 
> Review URL: https://webrtc-codereview.appspot.com/5049004

TBR=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5285 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 16:16:45 +00:00
..
build
Move realtime tests to webrtc_perf_tests.
2013-12-13 12:48:05 +00:00
common_audio
Roll chromium_revision 232627:238260
2013-12-11 13:37:12 +00:00
common_video
Add SwapFrame() to VideoSendStreamInput.
2013-12-11 16:26:16 +00:00
examples/android
Complete rewrite of demo application.
2013-12-12 16:55:37 +00:00
modules
Revert 5228 "Use the RTT from RtcpRttStats class if provided whe..."
2013-12-13 16:16:45 +00:00
system_wrappers
Roll chromium_revision 232627:238260
2013-12-11 13:37:12 +00:00
test
Update talk to 58127566 together with
2013-12-13 00:21:03 +00:00
tools
Roll chromium_revision 232627:238260
2013-12-11 13:37:12 +00:00
video
Move realtime tests to webrtc_perf_tests.
2013-12-13 12:48:05 +00:00
video_engine
Callback for send bitrate estimates - new roll
2013-12-13 09:46:59 +00:00
voice_engine
Enables mixing and matching Java and native audio. It is used for getting best of both worlds capabilities (AEC and low latency).
2013-12-11 21:42:44 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
call.h
Adds support for sending redundant payloads over RTX.
2013-12-04 10:24:26 +00:00
common_types.h
Callback for send bitrate estimates - new roll
2013-12-13 09:46:59 +00:00
common.h
Add a Config class interface to AudioProcessing for passing options.
2013-07-25 18:28:29 +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
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
Correctly define OVERRIDE when building with g++ 4.7 and C++11 support
2013-12-10 19:20:46 +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
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
Add SwapFrame() to VideoSendStreamInput.
2013-12-11 16:26:16 +00:00
webrtc_examples.gyp
Complete rewrite of demo application.
2013-12-12 16:55:37 +00:00
webrtc_perf_tests.isolate
Move realtime tests to webrtc_perf_tests.
2013-12-13 12:48:05 +00:00
webrtc_tests.gypi
Merge metrics_unittests into video_engine_tests.
2013-12-13 14:31:47 +00:00
webrtc.gyp
Add SwapFrame() to VideoSendStreamInput.
2013-12-11 16:26:16 +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.