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nisse cae45d0469 Move RtpTransportControllerSend to a new file. 
Also move RtpTransportControllerSendInterface to its own header file.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2808043002
Cr-Commit-Position: refs/heads/master@{#17840}
2017-04-24 12:53:20 +00:00
..
api
Enable GN check for webrtc/{p2p,system_wrappers}
2017-04-21 12:17:08 +00:00
audio
Move RtpTransportControllerSend to a new file.
2017-04-24 12:53:20 +00:00
base
Negotiate the same SRTP crypto suites for every DTLS association formed.
2017-04-21 10:23:33 +00:00
call
Move RtpTransportControllerSend to a new file.
2017-04-24 12:53:20 +00:00
common_audio
Enable GN check for webrtc/base
2017-04-19 15:37:36 +00:00
common_video
Delete unneeded includes of deprecated system_wrappers include files.
2017-03-30 07:31:15 +00:00
examples
Espresso test for loopback video quality testing
2017-04-24 11:15:13 +00:00
logging
Delay based logging.
2017-04-11 08:50:23 +00:00
media
Add content type extension to capabilities
2017-04-24 12:12:35 +00:00
modules
Reland of GN: Enable ARC for Mac and iOS in rtc_* templates (patchset #1 id:1 of https://codereview.webrtc.org/2827223003/ )
2017-04-24 07:57:16 +00:00
ortc
Relanding: Remove rtc_p2p_unittests from ortc_unittests and rtc_media_unittests
2017-04-18 22:49:09 +00:00
p2p
Fixing SignalSentPacket for TCP connections.
2017-04-21 21:22:23 +00:00
pc
Negotiate the same SRTP crypto suites for every DTLS association formed.
2017-04-21 10:23:33 +00:00
sdk
Reland of GN: Enable ARC for Mac and iOS in rtc_* templates (patchset #1 id:1 of https://codereview.webrtc.org/2827223003/ )
2017-04-24 07:57:16 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Enable GN check for webrtc/{p2p,system_wrappers}
2017-04-21 12:17:08 +00:00
test
Reland of GN: Enable ARC for Mac and iOS in rtc_* templates (patchset #1 id:1 of https://codereview.webrtc.org/2827223003/ )
2017-04-24 07:57:16 +00:00
tools
Add network tester client [android] to be able to test mobile networks in terms of packet size and sending rates.
2017-04-20 13:56:27 +00:00
video
Move RtpTransportControllerSend to a new file.
2017-04-24 12:53:20 +00:00
voice_engine
Move RtpTransportControllerSend to a new file.
2017-04-24 12:53:20 +00:00
.gitignore
…
BUILD.gn
Reland of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #1 id:1 of https://codereview.webrtc.org/2786363002/ )
2017-04-07 07:59:12 +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
Add read support of RtpStreamId/RepairedRtpStreamId header extensions.
2017-04-19 09:59:48 +00:00
config.cc
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
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
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
OWNERS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +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_decoder.h
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +00:00
video_encoder.h
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +00:00
video_frame.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
video_receive_stream.h
Delete VieRemb class, move functionality to PacketRouter.
2017-04-19 06:38:35 +00:00
video_send_stream.h
Add number of quality adapt changes to VideoSendStream stats.
2017-04-04 12:51:49 +00:00
webrtc.gni
Reland of GN: Enable ARC for Mac and iOS in rtc_* templates (patchset #1 id:1 of https://codereview.webrtc.org/2827223003/ )
2017-04-24 07:57:16 +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.