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VladimirTechMan 7b3eca9940 Add media constraint-key constants for generating offers and answers 
Similar to the existing constants for the media constraint-keys intended
for media sources, add the constants for the standard keys to generate
offers and answers.

This patch also adds a few comments to RTCMediaConstraints.h, to give
a better clue on the intended usage scope of declared media-constraint
keys and values.

BUG=webrtc:7722

Review-Url: https://codereview.webrtc.org/2908013002
Cr-Commit-Position: refs/heads/master@{#18362}
2017-06-01 01:25:48 +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
Support epoll in PhysicalSocketServer.
2017-05-31 20:09:18 +00:00
call
Create unit tests for RtpDemuxer
2017-05-31 16:12:25 +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
Use oneof in event log proto to reduce memory consumption.
2017-05-31 12:45:56 +00:00
media
Recreate FlexfecReceiveStream separately from VideoReceiveStream.
2017-05-30 08:31:37 +00:00
modules
Small cleanup of rtp_rtcp testAPI tests.
2017-05-31 14:35:16 +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
Add media constraint-key constants for generating offers and answers
2017-06-01 01:25:48 +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
Propagate input capture time in ViEEncoder::OnFrame.
2017-05-31 13:33:21 +00:00
tools
Ensure the RtcEventLog parser is backwards compatible
2017-05-31 09:03:16 +00:00
video
Disable flaky test EndToEndTest.TestFlexfecRtpStatePreservation on linux for now.
2017-05-31 14:46:56 +00:00
voice_engine
This cl removes RtcEventLog deps to call:call_interfaces. The purpose is to be able to use the event log from the upcoming RtpTransport.
2017-05-30 10:52:10 +00:00
.gitignore
…
BUILD.gn
Create unit tests for RtpDemuxer
2017-05-31 16:12:25 +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
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
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
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.