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platform-external-webrtc/webrtc
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sakal 68b5df97c7 Add video codec setting to AppRTCMobile on iOS. 
List items in settings menu no longer stay selected. Checkmarks are now
added to the selected options before the view appears.

BUG=webrtc:7316
TBR=denicija

Review-Url: https://codereview.webrtc.org/2735303004
Cr-Commit-Position: refs/heads/master@{#17296}
2017-03-17 16:01:59 +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
Add video codec setting to AppRTCMobile on iOS.
2017-03-17 16:01:59 +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
Delete unused test code in modules/video_coding/test/
2017-03-17 15:11:11 +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
Minor FlexFEC fixes.
2017-03-17 14:02:46 +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
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
(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
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
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
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
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.