admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc
History
philipel c18f1d7c94 Revert of Fix off-by-one bugs in video_coding::PacketBuffer when the buffer is filled with a single frame. (patchset #5 id:80001 of https://codereview.chromium.org/2993513002/ ) 
Reason for revert:
Break performance bots.

Original issue's description:
> Fix off-by-one bugs in video_coding::PacketBuffer when the buffer is filled with a single frame.
>
> BUG=webrtc:8028
>
> Review-Url: https://codereview.webrtc.org/2993513002
> Cr-Commit-Position: refs/heads/master@{#19209}
> Committed: ee13e8919c

TBR=stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:8028

Review-Url: https://codereview.webrtc.org/2990183002
Cr-Commit-Position: refs/heads/master@{#19211}
2017-08-02 11:18:02 +00:00
..
api
Reland of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt
2017-08-01 06:22:01 +00:00
audio
Reland "Allow AudioSendStream to reconfig AudioNetworkAdaptor"
2017-07-26 14:28:51 +00:00
base
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
call
Only one implementation of MockRtpPacketSink once
2017-07-28 17:05:45 +00:00
common_audio
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
common_video
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
examples
Update native plugin dll for turn servers and video.
2017-07-25 23:04:31 +00:00
logging
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
media
Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public.
2017-07-26 09:09:44 +00:00
modules
Revert of Fix off-by-one bugs in video_coding::PacketBuffer when the buffer is filled with a single frame. (patchset #5 id:80001 of https://codereview.chromium.org/2993513002/ )
2017-08-02 11:18:02 +00:00
ortc
Reland of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt
2017-08-01 06:22:01 +00:00
p2p
Relanding: Move "max IPv6 networks" logic to BasicPortAllocator, and fix sorting.
2017-07-26 23:09:33 +00:00
pc
Reland of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt
2017-08-01 06:22:01 +00:00
rtc_base
If adapter type is unknown and interface name is "ipsec", treat as VPN.
2017-07-27 22:05:29 +00:00
rtc_tools
Print general usage information for event_log_analyzer
2017-07-20 15:05:09 +00:00
sdk
ObjC: Fix quality scaling for injected encoders
2017-08-01 15:07:59 +00:00
stats
Make RTCStatsReport::ToString() return JSON-parseable string.
2017-07-28 14:29:12 +00:00
system_wrappers
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
test
Rename isolated_output to test_output and add a method to get the test_output directory.
2017-07-25 14:31:18 +00:00
video
Explicitly inform PacketRouter which RTP-RTCP modules are REMB-candidates
2017-08-01 13:30:28 +00:00
voice_engine
Explicitly inform PacketRouter which RTP-RTCP modules are REMB-candidates
2017-08-01 13:30:28 +00:00
.gitignore
…
BUILD.gn
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
codereview.settings
…
common_types.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
common_types.h
Add parsing/serializing for MID RTP header extension.
2017-07-21 17:33:25 +00:00
config_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
DEPS
Move webrtc/{tools => rtc_tools}
2017-06-30 10:04:59 +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
…
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
Report interframe delay sum in old GetStats
2017-07-07 08:26:24 +00:00
video_send_stream.h
Move RTP keep-alive config from VideoSendStream::Config to Call::Config
2017-07-11 10:44:17 +00:00
webrtc.gni
Turning on Opus 120ms frame length switch.
2017-07-27 17:23:35 +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.