admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc
History
andresp@webrtc.org 44eb87e6dd Tool to establish a loopback call via apprtc turn server. 
For now the test keeps track of video bandwidth estimation and plots it
using google visualization libraries after the test is concluded.
There is also scripts to run a test and record the tcpdump.

BUG=3037
R=hta@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5707 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-17 14:23:22 +00:00
..
build
Enable VS2013 for Windows compilation by default.
2014-03-12 14:41:58 +00:00
common_audio
Add a float interface to PushSincResampler.
2014-03-10 18:51:42 +00:00
common_video
Add an AlignedFreeDeleter and remove scoped_ptr_malloc.
2014-02-20 21:08:36 +00:00
examples
Android, WebRTCDemo: fixes crash issue when pressing switch camera button on devices with only one camera.
2014-01-24 18:42:12 +00:00
modules
References to includes in third_party should be relative, not absolute.
2014-03-14 16:22:18 +00:00
system_wrappers
Remove std:: prefixes from C functions in webrtc/.
2014-03-07 15:23:34 +00:00
test
Implement minimum transmit bitrate.
2014-03-13 12:52:27 +00:00
tools
Tool to establish a loopback call via apprtc turn server.
2014-03-17 14:23:22 +00:00
video
Stopping network threads before tearing down test
2014-03-13 15:39:27 +00:00
video_engine
Implement minimum transmit bitrate.
2014-03-13 12:52:27 +00:00
voice_engine
Voice Engine GetRemoteCSRCs should return the CSRCs from rtp_receiver_ instead of _rtpRtcpModule now.
2014-03-11 16:19:56 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
call.h
Add configuration for cpu overuse detection to video send stream.
2014-01-31 10:05:07 +00:00
common_types.h
Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR).
2014-02-19 11:59:02 +00:00
common.h
Add a Config class interface to AudioProcessing for passing options.
2013-07-25 18:28:29 +00:00
config.h
Wire up statistics in video receive stream of new API
2014-02-07 12:06:29 +00:00
engine_configurations.h
Remove external encryption API for VoE.
2014-02-18 11:27:22 +00:00
experiments.h
Missing include in experiments.h
2014-02-25 09:17:43 +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
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
supplement.gypi
Removes script for generating supplement.gypi also adds git ignore for tools/gn.
2014-01-21 15:54:56 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Moves WEBRTC_POSIX define from header file to gyp-settings.
2014-03-07 15:30:21 +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
Make VideoReceiveStream::GetStats() const.
2014-02-07 15:32:45 +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
Re-landing "Routing SuspendChange to VideoSendStream::Stats"
2014-03-13 13:31:21 +00:00
webrtc_examples.gyp
Android example apps: fixes issue where useful failure information was suppressed.
2014-01-21 19:03:51 +00:00
webrtc_perf_tests.isolate
Move realtime tests to webrtc_perf_tests.
2013-12-13 12:48:05 +00:00
webrtc_tests.gypi
Adding NetEq performance test to webrtc_perf_tests
2014-01-10 08:24:04 +00:00
webrtc.gyp
Integrate fake_network_pipe into direct_transport.
2013-12-18 20:28:25 +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.