admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc
History
philipel 3bf97cf060 Workaround for PacketBuffer bug. 
There exist a bug in the video_coding::PacketBuffer which triggers when a
frame is the same size as the buffer. A trivial workaround is to increase
the start size to something big so that this never happens in practice.

The bug has been fixed but we still want to test the workaround in ToT,
which is why this CL exist.

BUG=webrtc:8028, chromium:752886
R=stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/2994093002 .
Cr-Commit-Position: refs/heads/master@{#19308}
2017-08-10 16:11:04 +00:00
..
api
Removing VCMCodecDataBase::Codec and VideoCodingModule::Codec.
2017-08-10 09:43:14 +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
Add functionality which limits the number of bytes on the network.
2017-08-09 14:17:33 +00:00
common_audio
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
common_video
Removing VCMCodecDataBase::Codec and VideoCodingModule::Codec.
2017-08-10 09:43:14 +00:00
examples
Add Android Camera To Unity Plugin
2017-08-09 00:08:03 +00:00
logging
Replace CHECK(x && y) with two separate CHECK() calls
2017-08-10 00:22:01 +00:00
media
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
modules
Add PacketRouter::SetMaxDesiredReceiveBitrate for application limited receive bandwidth
2017-08-10 13:03:57 +00:00
ortc
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
p2p
Replace CHECK(x && y) with two separate CHECK() calls
2017-08-10 00:22:01 +00:00
pc
Replace CHECK(x && y) with two separate CHECK() calls
2017-08-10 00:22:01 +00:00
rtc_base
Replace CHECK(x && y) with two separate CHECK() calls
2017-08-10 00:22:01 +00:00
rtc_tools
Use default header extension map in rtc_event_log2text
2017-08-08 12:40:26 +00:00
sdk
Support Java VideoFrames in MediaCodecVideoEncoder.
2017-08-10 11:15:42 +00:00
stats
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
system_wrappers
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
test
Removing VCMCodecDataBase::Codec and VideoCodingModule::Codec.
2017-08-10 09:43:14 +00:00
video
Workaround for PacketBuffer bug.
2017-08-10 16:11:04 +00:00
voice_engine
Add thread annotations to AudioLevel
2017-08-10 12:01:21 +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
Wire up RTP keep-alive in ortc api.
2017-08-09 13:42:32 +00:00
config_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config.cc
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
config.h
Audit of kConstants missing the const qualifier
2017-08-08 17:48:15 +00:00
DEPS
Move webrtc/{tools => rtc_tools}
2017-06-30 10:04:59 +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
Protected streams report RTP messages directly to the FlexFec streams
2017-08-02 14:39:07 +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
Removing unused declared arg
2017-08-03 17:45:31 +00:00
whitespace.txt
…

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.