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platform-external-webrtc/webrtc
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denicija 8c375dedb2 Expose bit rate property in ARDAppClient and set max bitrate for video RTCRtcSender. 
BUG=webrtc:6654

Review-Url: https://codereview.webrtc.org/2484733002
Cr-Commit-Position: refs/heads/master@{#14977}
2016-11-08 14:28:25 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Added audio mixer and removed audio device module in AudioState::Config.
2016-11-08 12:26:37 +00:00
audio
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +00:00
base
Update Visual Studio runtime file copying
2016-11-03 18:26:40 +00:00
build
GN: Move xmllite_xmpp_unittests into webrtc/libjingle
2016-11-08 11:18:43 +00:00
call
Remove audio from BitrateEstimatorTest.
2016-11-08 11:45:01 +00:00
common_audio
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
2016-10-17 06:56:20 +00:00
common_video
Add function for getting supported H264 level from max resolution and fps
2016-11-08 10:57:58 +00:00
examples
Expose bit rate property in ARDAppClient and set max bitrate for video RTCRtcSender.
2016-11-08 14:28:25 +00:00
libjingle
GN: Move xmllite_xmpp_unittests into webrtc/libjingle
2016-11-08 11:18:43 +00:00
logging
GN: New conventions, default target and refactorings
2016-10-28 12:44:07 +00:00
media
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +00:00
modules
Disabling NOTREACHED which we're hitting flakily in browser tests.
2016-11-08 13:32:07 +00:00
p2p
Moving stun_prober target from webrtc/p2p to webrtc/examples
2016-11-01 10:09:19 +00:00
pc
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +00:00
sdk
iOS: Add improved BWE field trial.
2016-11-07 19:39:01 +00:00
stats
RTCInboundRTPStreamStats[1] added.
2016-11-01 10:00:24 +00:00
system_wrappers
Add stats for frequency offset when converting RTP timestamp to NTP time.
2016-11-02 14:14:10 +00:00
test
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +00:00
tools
GN: New conventions, default target and refactorings
2016-10-28 12:44:07 +00:00
video
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +00:00
voice_engine
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +00:00
.gitignore
…
BUILD.gn
GN: Move xmllite_xmpp_unittests into webrtc/libjingle
2016-11-08 11:18:43 +00:00
call.h
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +00:00
codereview.settings
…
common_types.cc
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
common_types.h
Add new codec for FlexFEC.
2016-11-07 11:04:03 +00:00
common.gyp
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
config.cc
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +00:00
config.h
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +00:00
DEPS
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete talk directory, and references from build_ios_libs.sh.
2016-09-02 10:07:13 +00:00
OWNERS
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
supplement.gypi
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +00:00
transport.h
…
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
video_encoder.h
Remove deprected functions from EncodedImageCallback and RtpRtcp
2016-11-04 18:39:37 +00:00
video_frame.h
Revert of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #7 id:120001 of https://codereview.webrtc.org/2383093002/ )
2016-11-02 10:40:05 +00:00
video_receive_stream.h
Implement framesDecoded stat in video receive ssrc stats.
2016-10-26 14:09:29 +00:00
video_send_stream.h
Let ViEEncoder express resolution requests as Sinkwants.
2016-11-01 18:45:54 +00:00
voice_engine_configurations.h
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
webrtc_examples.gyp
Moving stun_prober target from webrtc/p2p to webrtc/examples
2016-11-01 10:09:19 +00:00
webrtc_tests.gypi
GYP: Remove targets inside include_tests==1 that are converted to GN.
2016-09-15 11:57:39 +00:00
webrtc.gyp
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +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.