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webrtc_m130/webrtc
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michaelt 333d0ff631 Add first part of the network_tester functionality. 
BUG=webrtc:7426

Review-Url: https://codereview.webrtc.org/2779233002
Cr-Commit-Position: refs/heads/master@{#17635}
2017-04-11 06:26:35 +00:00
..
api
Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ )
2017-04-10 14:39:05 +00:00
audio
Making FakeNetworkPipe demux audio and video packets.
2017-04-10 23:57:57 +00:00
base
Make safe_cmp::* constexpr
2017-04-11 05:44:07 +00:00
call
Making FakeNetworkPipe demux audio and video packets.
2017-04-10 23:57:57 +00:00
common_audio
Rename COMPILE_ASSERT macro to RTC_COMPILE_ASSERT
2017-04-11 06:21:43 +00:00
common_video
Delete unneeded includes of deprecated system_wrappers include files.
2017-03-30 07:31:15 +00:00
examples
Removing unnecessary parameters from initializeAndroidGlobals.
2017-04-10 22:08:02 +00:00
logging
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
media
Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ )
2017-04-10 14:39:05 +00:00
modules
Rename COMPILE_ASSERT macro to RTC_COMPILE_ASSERT
2017-04-11 06:21:43 +00:00
ortc
Disable the ORTC integration tests on TSan.
2017-03-28 20:45:18 +00:00
p2p
Fix compilation issues of std::unique_ptr
2017-04-09 16:09:06 +00:00
pc
Make RtpTransport actually implement RtpTransportInterface
2017-04-10 23:17:57 +00:00
sdk
Restore old (deprecated) signature of initializeAndroidGlobals.
2017-04-11 01:07:55 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Rename COMPILE_ASSERT macro to RTC_COMPILE_ASSERT
2017-04-11 06:21:43 +00:00
test
Making FakeNetworkPipe demux audio and video packets.
2017-04-10 23:57:57 +00:00
tools
Add first part of the network_tester functionality.
2017-04-11 06:26:35 +00:00
video
Making FakeNetworkPipe demux audio and video packets.
2017-04-10 23:57:57 +00:00
voice_engine
Removing unnecessary parameters from initializeAndroidGlobals.
2017-04-10 22:08:02 +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
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Revert of Supporting 48kHz PCM file. (patchset #1 id:1 of https://codereview.webrtc.org/2790493004/ )
2017-03-31 23:32:28 +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
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
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
video_receive_stream.h
Add |protected_by_flexfec| flag to VideoReceiveStream::Config.
2017-03-09 14:13:20 +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 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.