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brandtr 48d21a23c6 Persist RTP state for FlexFEC. 
Before this CL, the RTP state would be re-randomized after a
recreation of VideoSendStream. That might lead to us sending
a non-compliant RTP stream, which is avoided after the
changes in this CL.

BUG=webrtc:5654
TBR=pbos@webrtc.org  # Trivial change to fuzzer.

Review-Url: https://codereview.webrtc.org/2912713002
Cr-Commit-Position: refs/heads/master@{#18322}
2017-05-30 09:32:12 +00:00
..
api
Add separate base classes for I420 and I444 buffers
2017-05-30 08:21:59 +00:00
audio
Rename elad.alon to eladalon, to avoid confusion between repositories.
2017-05-25 07:15:35 +00:00
base
Disable SIGPIPE for sockets created on iOS.
2017-05-26 23:27:09 +00:00
call
Store/restore RTP state for audio streams with same SSRC within a call
2017-05-23 13:07:11 +00:00
common_audio
Moving compile_assert_c.h to webrtc/base
2017-05-15 15:06:47 +00:00
common_video
Add separate base classes for I420 and I444 buffers
2017-05-30 08:21:59 +00:00
examples
Reland of Split iOS sdk in to separate targets (patchset #1 id:1 of https://codereview.webrtc.org/2911053002/ )
2017-05-30 08:48:47 +00:00
logging
Remove final from RtcEventLogNullImpl
2017-05-29 09:46:05 +00:00
media
Recreate FlexfecReceiveStream separately from VideoReceiveStream.
2017-05-30 08:31:37 +00:00
modules
Persist RTP state for FlexFEC.
2017-05-30 09:32:12 +00:00
ortc
Remove VirtualSocketServer's dependency on PhysicalSocketServer.
2017-05-17 01:00:06 +00:00
p2p
Fix issue with send-side bandwidth estimation over TURN TCP connections.
2017-05-27 01:40:05 +00:00
pc
Reland of Removes usage of native base::android::GetApplicationContext()
2017-05-26 08:51:53 +00:00
sdk
Reland of Adding backward compatibility header (patchset #1 id:1 of https://codereview.webrtc.org/2908143003/ )
2017-05-30 08:50:35 +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
Persist RTP state for FlexFEC.
2017-05-30 09:32:12 +00:00
tools
Revert of Remove unecessary non fatal error statement that very often is printed in the PSNR or SSIM metric n… (patchset #1 id:1 of https://codereview.webrtc.org/2901793002/ )
2017-05-23 15:31:11 +00:00
video
Persist RTP state for FlexFEC.
2017-05-30 09:32:12 +00:00
voice_engine
Delete RtpData::OnRecoveredPacket, use RecoveredPacketReceiver instead.
2017-05-29 15:16:37 +00:00
.gitignore
…
BUILD.gn
Reland of Split iOS sdk in to separate targets (patchset #1 id:1 of https://codereview.webrtc.org/2911053002/ )
2017-05-30 08:48:47 +00:00
call.h
…
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
…
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
Reland of Removes usage of native base::android::GetApplicationContext()
2017-05-26 08:51:53 +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.