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webrtc_m130/webrtc
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zhihuang b2cdd93fd6 Remove the dependency of TransportChannel and TransportChannelImpl. 
DtlsTransportChannelWrapper is renamed to be DtlsTransport which inherits from
DtlsTransportInternal. There will be no concept of "channel" in p2p level.
Both P2PTransportChannel and DtlsTransport don't depend on TransportChannel
and TransportChannelImpl any more and they are removed in this CL.

BUG=none

Review-Url: https://codereview.webrtc.org/2606123002
Cr-Commit-Position: refs/heads/master@{#16173}
2017-01-20 00:54:25 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Remove the dependency of TransportChannel and TransportChannelImpl.
2017-01-20 00:54:25 +00:00
audio
Pass SdpAudioFormat through Channel, without converting to CodecInst
2017-01-19 15:03:59 +00:00
base
Reduce the log verbosity in sslstreamadapter_unittest
2017-01-18 23:29:35 +00:00
build
GN: Refactor modules_unittests to eliminate package boundary violations.
2017-01-19 16:27:11 +00:00
call
Pass SdpAudioFormat through Channel, without converting to CodecInst
2017-01-19 15:03:59 +00:00
common_audio
Replace RTC_DCHECK(false) with RTC_NOTREACHED().
2017-01-12 10:24:27 +00:00
common_video
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
examples
Fixing peerconnection reddish video issue
2017-01-18 12:23:10 +00:00
logging
GN: Refactor so that WebRTC compiles with rtc_enable_protobuf=false.
2017-01-17 10:37:34 +00:00
media
Remove the dependency of TransportChannel and TransportChannelImpl.
2017-01-20 00:54:25 +00:00
modules
GN: Add audio_conference_mixer_unittests to modules_unittests.
2017-01-19 19:57:56 +00:00
p2p
Remove the dependency of TransportChannel and TransportChannelImpl.
2017-01-20 00:54:25 +00:00
pc
Remove the dependency of TransportChannel and TransportChannelImpl.
2017-01-20 00:54:25 +00:00
sdk
Video collected by VideoFileRenderer is first saved on the native heap, then saved to disk during release.
2017-01-19 17:02:29 +00:00
stats
RTCMediaStreamTrackStats.kind added and collected.
2017-01-17 10:53:23 +00:00
system_wrappers
Reland of Delete unused code from systeminfo. (patchset #1 id:1 of https://codereview.webrtc.org/2584563004/ )
2017-01-09 09:22:16 +00:00
test
GN: Refactor modules_unittests to eliminate package boundary violations.
2017-01-19 16:27:11 +00:00
tools
Pass SdpAudioFormat through Channel, without converting to CodecInst
2017-01-19 15:03:59 +00:00
video
Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:230001 of https://codereview.webrtc.org/2642753002/ )
2017-01-19 08:06:17 +00:00
voice_engine
Pass SdpAudioFormat through Channel, without converting to CodecInst
2017-01-19 15:03:59 +00:00
.gitignore
…
BUILD.gn
Refactor OveruseFrameDetector to use timing in us units
2017-01-18 10:16:20 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
config.cc
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
config.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
DEPS
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +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
Remove extra uses of basictypes.h.
2016-12-09 14:54:08 +00:00
video_decoder.h
Move VideoDecoder::Create() logic to separate internal video decoder factory
2016-12-01 08:27:35 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
video_receive_stream.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
video_send_stream.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +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.