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ivoc e3fc11464e Fixed NetEq overflow bug. 
Negating an int can result in a value that cannot be represented as an int. This is fixed here by using a 64 bit variable.

BUG=chromium:663611

Review-Url: https://codereview.webrtc.org/2879863002
Cr-Commit-Position: refs/heads/master@{#18167}
2017-05-16 14:13:15 +00:00
..
api
Move webrtc/video_frame to common_video/include.
2017-05-15 09:42:11 +00:00
audio
New class RtpDemuxer and RtpPacketSinkInterface, use in Call.
2017-05-16 11:47:04 +00:00
base
Delete unused features of AsyncInvoke.
2017-05-16 12:51:29 +00:00
call
New class RtpDemuxer and RtpPacketSinkInterface, use in Call.
2017-05-16 11:47:04 +00:00
common_audio
Moving compile_assert_c.h to webrtc/base
2017-05-15 15:06:47 +00:00
common_video
Move webrtc/video_frame to common_video/include.
2017-05-15 09:42:11 +00:00
examples
Split iOS sdk in to separate targets
2017-05-16 14:06:59 +00:00
logging
DCHECK that we don't insert nullptr into event log.
2017-05-11 13:05:27 +00:00
media
WebRtcVideoEncoderFactory cleanup
2017-05-12 07:24:40 +00:00
modules
Fixed NetEq overflow bug.
2017-05-16 14:13:15 +00:00
ortc
Delete method MessageQueue::set_socketserver
2017-05-08 12:25:41 +00:00
p2p
Get tests working on systems that only support IPv6.
2017-05-16 02:43:33 +00:00
pc
Get tests working on systems that only support IPv6.
2017-05-16 02:43:33 +00:00
sdk
Split iOS sdk in to separate targets
2017-05-16 14:06:59 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Adding target to track asm_defines.h
2017-05-15 21:59:17 +00:00
test
Split iOS sdk in to separate targets
2017-05-16 14:06:59 +00:00
tools
Update testing tools (AppRTC, Go) to new versions
2017-05-16 12:36:15 +00:00
video
New class RtpDemuxer and RtpPacketSinkInterface, use in Call.
2017-05-16 11:47:04 +00:00
voice_engine
Resolves race between Channel::ProcessAndEncodeAudio() and Channel::StopSend()
2017-05-03 15:30:15 +00:00
.gitignore
…
BUILD.gn
Split iOS sdk in to separate targets
2017-05-16 14:06:59 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
…
common_types.cc
Add read support of RtpStreamId/RepairedRtpStreamId header extensions.
2017-04-19 09:59:48 +00:00
common_types.h
Delete RawVideoType enum, use the VideoType enum instead.
2017-04-28 14:18:05 +00:00
config.cc
Fix default extension id for video content type extension
2017-04-26 09:35:31 +00:00
config.h
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
2017-04-11 17:34:31 +00:00
DEPS
Move webrtc/video_frame to common_video/include.
2017-05-15 09:42:11 +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_frame.h
Move webrtc/video_frame to common_video/include.
2017-05-15 09:42:11 +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
Update adaptation stats to support degradations in both resolution and framerate.
2017-05-16 06:40:18 +00:00
webrtc.gni
Remove the expat and jsoncpp dependencies to reduce binary size.
2017-04-24 19:05:06 +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.