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zhihuang 243663965a Add the function ToRtpParameters. 
The function converts the cricket::Codec, cricket::RtpHeaderExtensions
and cricket::StreamParamsVec to webrtc::RtpParameters.

BUG=webrtc:7311

Review-Url: https://codereview.webrtc.org/2735853004
Cr-Commit-Position: refs/heads/master@{#17124}
2017-03-09 01:15:06 +00:00
..
api
Delete deprecated PeerConnection methods, and corresponding using declarations.
2017-03-08 14:59:45 +00:00
audio
Adding placeholder for low bandwidth audio test
2017-03-06 12:01:16 +00:00
base
Revert of Delete unused TaskRunner abstraction. (patchset #2 id:20001 of https://codereview.webrtc.org/2622923002/ )
2017-03-08 10:01:07 +00:00
call
Add support for Location (RTC_FROM_HERE) to ProcessThread::RegisterModule.
2017-03-03 11:20:24 +00:00
common_audio
Avoid overflow in WebRtcSpl_DotProductWithScale
2017-03-07 12:02:47 +00:00
common_video
Ignore aud and sei NALus when parsing bitstream.
2017-03-06 10:49:36 +00:00
examples
Revert of Add Metal video view in AppRTCMobile and metal availability macro. (patchset #5 id:80001 of https://codereview.webrtc.org/2722583002/ )
2017-03-08 14:54:59 +00:00
logging
Enable GN check for webrtc/logging
2017-03-08 10:12:11 +00:00
media
Refactor + enable GN check on video_coding_utility
2017-03-08 13:42:26 +00:00
modules
Conversational Speech generator, main script with shell arguments
2017-03-08 14:12:23 +00:00
ortc
Add the function ToRtpParameters.
2017-03-09 01:15:06 +00:00
p2p
Add the option to disable IPv6 ICE candidates on WiFi.
2017-03-07 22:40:51 +00:00
pc
Delete deprecated PeerConnection methods, and corresponding using declarations.
2017-03-08 14:59:45 +00:00
sdk
Delete deprecated PeerConnection methods, and corresponding using declarations.
2017-03-08 14:59:45 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Replace Clock::CurrentNtp with Clock::CurrentNtpTime
2017-03-07 10:51:09 +00:00
test
Remove VoEVolumeControl interface.
2017-03-08 09:52:20 +00:00
tools
GetTransportFeedbackVector return vector with lost packets too, sorted by seq-num
2017-03-08 13:03:53 +00:00
video
Disable failing 15 thumbernails full-stack test on windows
2017-03-08 09:17:35 +00:00
voice_engine
Remove VoEVolumeControl interface.
2017-03-08 09:52:20 +00:00
.gitignore
…
BUILD.gn
Remove voe_cmd_test.
2017-03-07 09:43:21 +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
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Remove the Windows Wave audio device implementation.
2017-02-17 22:48:07 +00:00
config.cc
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
config.h
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +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
no_op_function.cc
Moving no_op_function.cc out of webrtc/build
2017-01-24 13:25:12 +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
Only define NO_RETURN if undefined
2017-01-30 08:54:19 +00:00
video_decoder.h
Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
2017-02-09 12:53:45 +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
Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
2017-02-09 12:53:45 +00:00
video_send_stream.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
webrtc.gni
Reland of Setting is_component_build to false by default (patchset #1 id:1 of https://codereview.webrtc.org/2731703004/ )
2017-03-04 03:41:59 +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.