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zhihuang ebbe4f2ed5 Set the preferred DSCP value for Rtp data channel to be DSCP_AF41. 
BUG=b/31996729

Review-Url: https://codereview.webrtc.org/2539813003
Cr-Commit-Position: refs/heads/master@{#15449}
2016-12-06 18:45:47 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Set the preferred DSCP value for Rtp data channel to be DSCP_AF41.
2016-12-06 18:45:47 +00:00
audio
Move functionality out from AudioFrame and into AudioFrameOperations.
2016-12-05 09:46:20 +00:00
base
Use RTC_DISALLOW_COPY_AND_ASSIGN in webrtc/base/sigslottester.h
2016-12-05 14:14:34 +00:00
build
Move /webrtc/api/android files to /webrtc/sdk/android
2016-12-06 12:29:45 +00:00
call
Revert "Revert of Use different restrictions of acked bitrate lag depending on operating point. (patchset #3 id:40001 of https://codereview.webrtc.org/2542083003/ )"
2016-12-05 16:23:48 +00:00
common_audio
Split targets mixing .c and .cc sources.
2016-12-06 06:47:52 +00:00
common_video
Sanity check parsed QP values from H264 bitstream
2016-12-02 09:29:53 +00:00
examples
Move /webrtc/api/android files to /webrtc/sdk/android
2016-12-06 12:29:45 +00:00
libjingle
Remove all references to GYP
2016-11-16 19:11:38 +00:00
logging
Replace test_support_main by test_main and get rid of test_support_main_threaded_mac
2016-11-30 14:12:10 +00:00
media
Set the preferred DSCP value for Rtp data channel to be DSCP_AF41.
2016-12-06 18:45:47 +00:00
modules
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
p2p
Remove all references to GYP
2016-11-16 19:11:38 +00:00
pc
Set the preferred DSCP value for Rtp data channel to be DSCP_AF41.
2016-12-06 18:45:47 +00:00
sdk
Add unit tests for avfoundation format mapper functions.
2016-12-06 13:22:53 +00:00
stats
Collecting RTCIceCandidatePairStats.transport_id and improved unittests.
2016-11-30 09:50:36 +00:00
system_wrappers
Add comment to metrics.h
2016-12-06 11:59:08 +00:00
test
Reland of H.264 packetization mode 0 (try 3) (patchset #1 id:1 of https://codereview.webrtc.org/2558453002/ )
2016-12-06 13:36:13 +00:00
tools
Added tool for reference less video analysis (go/refless-video-analysis)
2016-12-02 13:00:10 +00:00
video
Reland of H.264 packetization mode 0 (try 3) (patchset #1 id:1 of https://codereview.webrtc.org/2558453002/ )
2016-12-06 13:36:13 +00:00
voice_engine
Move WEBRTC_VOICE_ENGINE_TYPING_DETECTION to transmit_mixer.h
2016-12-05 10:22:18 +00:00
.gitignore
…
BUILD.gn
Add unit tests for avfoundation format mapper functions.
2016-12-06 13:22:53 +00:00
call.h
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +00:00
codereview.settings
…
common_types.cc
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabled.
2016-12-02 16:58:02 +00:00
config.cc
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
config.h
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
DEPS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
OWNERS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
transport.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
Move VideoDecoder::Create() logic to separate internal video decoder factory
2016-12-01 08:27:35 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ )
2016-11-11 11:55:19 +00:00
video_receive_stream.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
video_send_stream.h
Implement periodic bandwidth probing in application-limited region.
2016-11-28 21:11:24 +00:00
voice_engine_configurations.h
Move WEBRTC_VOICE_ENGINE_TYPING_DETECTION to transmit_mixer.h
2016-12-05 10:22:18 +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.