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webrtc/webrtc
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henrike@webrtc.org 6a782c2a46 Revert 7693 "Add jmi field for packets discarded due to network error" breaks chromium's webrtc_cases. 
TBR=guoweis@webrtc.org

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/25179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7706 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-14 22:33:13 +00:00
..
base
Use ScreenCapturer to capture the whole and clip to the window rect when the shared window is on the top.
2014-11-11 18:15:55 +00:00
build
Add VP9 codec to VCM and vie_auto_test.
2014-11-01 06:10:48 +00:00
common_audio
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
common_video
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
examples
Update Android projects to API level 21.
2014-10-31 23:26:10 +00:00
libjingle
move xmpp and p2p to webrtc
2014-10-28 22:20:11 +00:00
modules
Wrap the splitting filter in its own class
2014-11-14 22:18:10 +00:00
overrides
…
p2p
Revert 7693 "Add jmi field for packets discarded due to network error" breaks chromium's webrtc_cases.
2014-11-14 22:33:13 +00:00
sound
rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation.
2014-10-01 16:33:03 +00:00
system_wrappers
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
test
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
tools
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video
Disable EndToEnd.GetStats test.
2014-11-14 17:42:51 +00:00
video_engine
Report total bitrate for all streams in GetStats.
2014-11-14 11:52:04 +00:00
voice_engine
Report total bitrate for all streams in GetStats.
2014-11-14 11:52:04 +00:00
.gitignore
…
BUILD.gn
Change the static_library("webrtc") to a source set in the GN build.
2014-11-12 07:56:21 +00:00
call.h
Wire up bandwidth stats to the new API and webrtcvideoengine2.
2014-11-05 14:05:29 +00:00
common_types.h
Wire up bandwidth stats to the new API and webrtcvideoengine2.
2014-11-05 14:05:29 +00:00
common.gyp
…
common.h
…
config.cc
Log formatting fix for VideoEncoderConfig.
2014-11-06 09:35:08 +00:00
config.h
Wire up bandwidth stats to the new API and webrtcvideoengine2.
2014-11-05 14:05:29 +00:00
engine_configurations.h
Add VP9 codec to VCM and vie_auto_test.
2014-11-01 06:10:48 +00:00
experiments.h
…
frame_callback.h
…
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
…
PATENTS
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PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
supplement.gypi
…
transport.h
…
typedefs.h
…
video_decoder.h
Add support for VP9 in webrtc::Call and video_loopback.
2014-11-04 19:41:15 +00:00
video_encoder.h
Add VP9 codec to VCM and vie_auto_test.
2014-11-01 06:10:48 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
…
video_receive_stream.h
Wire up bandwidth stats to the new API and webrtcvideoengine2.
2014-11-05 14:05:29 +00:00
video_renderer.h
…
video_send_stream.h
Wire up bandwidth stats to the new API and webrtcvideoengine2.
2014-11-05 14:05:29 +00:00
webrtc_examples.gyp
Add macros and APIs for webrtc histograms.
2014-10-23 12:57:56 +00:00
webrtc_perf_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
webrtc_tests.gypi
Put send-side bwe probing under finch experiment.
2014-11-10 13:55:16 +00:00
webrtc.gyp
Add VP9 codec to VCM and vie_auto_test.
2014-11-01 06:10:48 +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.