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zhihuang 86abd6f2fd Add an abstract class for IceTransport 
P2PTransportChannl will be renamed to IceTransport and this class will be
the base class of IceTransport. By doing this, the Dtls related methods
can be separated from the IceTransport.

For more detail, https://docs.google.com/document/d/1g9RA0s4RV7hFAcWiAM2b6H5ZohAVpBqcYienEDj6IcY/edit

BUG=none

Review-Url: https://codereview.webrtc.org/2577183004
Cr-Commit-Position: refs/heads/master@{#15690}
2016-12-19 19:54:05 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Removed undefined method from webrtcsession.h.
2016-12-19 17:34:14 +00:00
audio
Add ossu@ to OWNERS of audio/ and modules/audio_coding/
2016-12-15 15:52:14 +00:00
base
Revert of Delete unused code from systeminfo. (patchset #3 id:40001 of https://codereview.webrtc.org/2578323005/ )
2016-12-17 00:48:28 +00:00
build
Move tools/mb -> tools-webrtc/mb
2016-12-18 21:14:50 +00:00
call
Move FlexfecReceiveStream from api/call/ to call/.
2016-12-19 09:13:46 +00:00
common_audio
Fix for left shift of negative value in NetEq.
2016-12-14 10:00:02 +00:00
common_video
Reland of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #1 id:1 of https://codereview.webrtc.org/2574123002/ )
2016-12-15 14:30:00 +00:00
examples
Add QP stats to the statsview in AppRTCMobile for ios.
2016-12-19 10:14:08 +00:00
libjingle
Remove extra uses of basictypes.h.
2016-12-09 14:54:08 +00:00
logging
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
media
Clean up storage of FlexFEC payload type in webrtc::VideoCodecSettings.
2016-12-19 17:41:04 +00:00
modules
Add GUARDED_BY's in FlexfecReceiver.
2016-12-19 18:02:30 +00:00
p2p
Add an abstract class for IceTransport
2016-12-19 19:54:05 +00:00
pc
Removing "crypto_required" from MediaContentDescription.
2016-12-13 19:29:16 +00:00
sdk
Add disabled certificate check support to IceServer PeerConnection API.
2016-12-19 12:10:30 +00:00
stats
Rename RTCOutboundRTPStreamStats *_rtt members to *_round_trip_time.
2016-12-15 14:17:15 +00:00
system_wrappers
Revert of Delete unused code from systeminfo. (patchset #3 id:40001 of https://codereview.webrtc.org/2578323005/ )
2016-12-17 00:48:28 +00:00
test
Add multithreaded fake encoder and corresponding FlexFEC VideoSendStreamTest.
2016-12-19 13:47:28 +00:00
tools
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
video
Revert of Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame. (patchset #1 id:1 of https://codereview.webrtc.org/2574943003/ )
2016-12-19 17:36:33 +00:00
voice_engine
Wire-up audio BWE with overhead.
2016-12-15 14:24:52 +00:00
.gitignore
…
BUILD.gn
Re-enable Opus complexity tests on Android
2016-12-19 14:18:35 +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
CodecInst operator<<
2016-12-19 14:04:04 +00:00
config.cc
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
config.h
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
DEPS
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +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
Reland of Properly report number of quality downscales in stats. (patchset #1 id:1 of https://codereview.webrtc.org/2586783003/ )
2016-12-19 14:32:16 +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
Implement periodic bandwidth probing in application-limited region.
2016-11-28 21:11:24 +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.