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webrtc_m130/webrtc
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terelius a2bb667a9b Use RTC_UNUSED instead of conditional compilation in BWE simulator tool. 
Mark ATTRIBUTE_UNUSED as deprecated since it only works with GCC and clang. I am not removing it now since typedefs.h is (perhaps incorrectly?) considered a public interface.

BUG=webrtc:7228

Review-Url: https://codereview.webrtc.org/2756483002
Cr-Commit-Position: refs/heads/master@{#17291}
2017-03-17 13:34:47 +00:00
..
api
Change the type of session_id() from string to int64_t.
2017-03-13 18:00:54 +00:00
audio
Move delay_based_bwe_ into CongestionController
2017-03-09 14:40:08 +00:00
base
Revert of Reland: Use native (optimized) functions for byte order conversion. (patchset #5 id:80001 of https://codereview.webrtc.org/2751403003/ )
2017-03-17 09:28:28 +00:00
call
Reland of write frame generator capturer to use TaskQueue instead of EventTimeWrapper (patchset #1 id:1 of https://codereview.webrtc.org/2751063005/ )
2017-03-17 12:55:25 +00:00
common_audio
iSAC fix entropy coder: Recently added DCHECK could in fact trigger
2017-03-13 12:28:47 +00:00
common_video
Add FullStack test for simulcast screenshare mode.
2017-03-09 17:23:30 +00:00
examples
Try re-enabling two PeerConnectionClientTests:
2017-03-15 12:50:13 +00:00
logging
Enable GN check for webrtc/logging
2017-03-08 10:12:11 +00:00
media
(Re)move VoE specific enums from common_types.h.
2017-03-16 08:20:23 +00:00
modules
Use RTC_UNUSED instead of conditional compilation in BWE simulator tool.
2017-03-17 13:34:47 +00:00
ortc
Add the function ToRtpParameters.
2017-03-09 01:15:06 +00:00
p2p
Making candidate pool size behave as decided in JSEP.
2017-03-10 23:18:00 +00:00
pc
Making candidate pool size behave as decided in JSEP.
2017-03-10 23:18:00 +00:00
sdk
Better handling of error condition in MediaCodecVideoEncoder.
2017-03-15 12:53:14 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Delete utf_util_win.h.
2017-03-14 08:32:50 +00:00
test
Reland of write frame generator capturer to use TaskQueue instead of EventTimeWrapper (patchset #1 id:1 of https://codereview.webrtc.org/2751063005/ )
2017-03-17 12:55:25 +00:00
tools
Improve error handling for ffmpeg operations
2017-03-15 15:27:31 +00:00
video
Reland of write frame generator capturer to use TaskQueue instead of EventTimeWrapper (patchset #1 id:1 of https://codereview.webrtc.org/2751063005/ )
2017-03-17 12:55:25 +00:00
voice_engine
Remove VoEHardware interface.
2017-03-15 15:08:07 +00:00
.gitignore
…
BUILD.gn
Remove voe_cmd_test.
2017-03-07 09:43:21 +00:00
call.h
…
codereview.settings
…
common_types.cc
…
common_types.h
(Re)move VoE specific enums from common_types.h.
2017-03-16 08:20:23 +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
…
no_op_function.cc
Moving no_op_function.cc out of webrtc/build
2017-01-24 13:25:12 +00:00
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Use RTC_UNUSED instead of conditional compilation in BWE simulator tool.
2017-03-17 13:34:47 +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
…
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
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.