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webrtc_m130/webrtc
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zhihuang 696f8ca2fa Handle the PROTO_TSL when getting the protocol priority. 
This bug breaks the internal project.

TBR=deadbeef@webrtc.org, pthacher@webrtc.org
BUG=webrtc:7889

Review-Url: https://codereview.webrtc.org/2959993002
Cr-Commit-Position: refs/heads/master@{#18792}
2017-06-27 22:11:24 +00:00
..
api
Don't forget to support G722 stereo decoding
2017-06-26 11:19:43 +00:00
audio
Disable AudioBweIntegrationTest.NoBandwidthDropAfterDtx - it's flaky
2017-06-26 12:04:12 +00:00
base
Rename safe_cmp::{Eq,Ne,Lt,Le,Ge,Gt} to Safe{Eq,Ne,Lt,Le,Ge,Gt}
2017-06-26 08:31:31 +00:00
call
Reland of Periodically update codec bit/frame rate settings.
2017-06-27 14:06:52 +00:00
common_audio
Put attribute before function name instead of after, as required by GCC
2017-06-26 08:32:40 +00:00
common_video
Return WrappedI444Buffer in VP9Impl
2017-06-23 03:28:06 +00:00
examples
Notify delegates about audio glitches in real time
2017-06-27 09:43:27 +00:00
logging
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
media
Reland of Periodically update codec bit/frame rate settings.
2017-06-27 14:06:52 +00:00
modules
Attempt to reduce AUDIO_RECORD_START_STATE_MISMATCH error rate on Android.
2017-06-27 15:58:43 +00:00
ortc
Fixing bad use of std::sort in test method.
2017-06-27 14:52:50 +00:00
p2p
Handle the PROTO_TSL when getting the protocol priority.
2017-06-27 22:11:24 +00:00
pc
Remove unused "crypto_options_" field.
2017-06-26 11:07:52 +00:00
rtc_base
Add webrtc/rtc_base skeleton.
2017-06-21 12:50:22 +00:00
sdk
Enable libjingle_peerconnection_datachannelonly_so target.
2017-06-27 20:20:05 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Prevent data race in GetStaticInstance
2017-05-29 07:01:14 +00:00
test
Reland of Periodically update codec bit/frame rate settings.
2017-06-27 14:06:52 +00:00
tools
Add NetEq delay plotting to event_log_visualizer
2017-06-14 13:09:58 +00:00
video
Fix deadlock in webrtc_perf_tests
2017-06-27 14:21:01 +00:00
voice_engine
Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface.
2017-06-21 08:05:22 +00:00
.gitignore
…
BUILD.gn
Add webrtc/rtc_base skeleton.
2017-06-21 12:50:22 +00:00
codereview.settings
…
common_types.cc
Allow parsing empty RTCP TargetBitrate messages, but stop sending them.
2017-06-22 12:40:25 +00:00
common_types.h
Allow parsing empty RTCP TargetBitrate messages, but stop sending them.
2017-06-22 12:40:25 +00:00
config.cc
Implement timing frames.
2017-06-19 14:18:55 +00:00
config.h
Implement timing frames.
2017-06-19 14:18:55 +00:00
DEPS
Delete old include file webrtc/video_frame.h.
2017-06-13 13:05:05 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Don't add or rename files in webrtc/ and webrtc/api/ without a proper review
2017-05-08 13:01:54 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Make WARN_UNUSED_RESULT a no-op on gcc
2017-04-06 11:32:27 +00:00
video_receive_stream.h
Delete VieRemb class, move functionality to PacketRouter.
2017-04-19 06:38:35 +00:00
video_send_stream.h
Add field trial for balanced degradation preference.
2017-06-14 12:52:21 +00:00
webrtc.gni
Adds detection of audio glitches for playout on iOS (reland)
2017-06-13 16:00:18 +00:00
whitespace.txt
Moving whitespace file up by one folder
2017-01-24 13:19:42 +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.