admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc
History
eladalon cb83bdf01f Create RtcpDemuxer. Capabilities: 
1. Demux RTCP messages according to the sender-SSRC.
2. Demux RTCP messages according to the RSID (resolved to an SSRC, then compared to the sender-RTCP).
3. Allow listening in on all RTCP messages passing through the demuxer ("broadcast sinks").

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2943693003
Cr-Commit-Position: refs/heads/master@{#18763}
2017-06-26 12:56:34 +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
Create RtcpDemuxer. Capabilities:
2017-06-26 12:56:34 +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
Always reset the audio session configuration after a call.
2017-06-26 07:19:11 +00:00
logging
Support building WebRTC without audio and video.
2017-06-15 19:52:32 +00:00
media
Enable more unittests on iOS, and disable those that fail on simulator
2017-06-22 08:47:20 +00:00
modules
Don't forget to support G722 stereo decoding
2017-06-26 11:19:43 +00:00
ortc
Remove VirtualSocketServer's dependency on PhysicalSocketServer.
2017-05-17 01:00:06 +00:00
p2p
Enable SNI in ssl adapter.
2017-06-16 23:30:48 +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
Revert of Disable RTCVideoRenderFrameCallbackNV12 test (patchset #1 id:20001 of https://codereview.webrtc.org/2951273002/ )
2017-06-26 10:11:51 +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
Force value-initialization of array to avoid uninitialized values.
2017-06-22 10:57:11 +00:00
tools
Add NetEq delay plotting to event_log_visualizer
2017-06-14 13:09:58 +00:00
video
Disable FullStackTest.SimulcastFullHdOveruse on Mac
2017-06-25 20:50:03 +00:00
voice_engine
Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface.
2017-06-21 08:05:22 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
BUILD.gn
Add webrtc/rtc_base skeleton.
2017-06-21 12:50:22 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
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
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
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
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
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.