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platform-external-webrtc/webrtc
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kwiberg bfc7f02d79 ArrayView: Support compile-time constant sizes 
BUG=none

Review-Url: https://codereview.webrtc.org/2667383006
Cr-Commit-Position: refs/heads/master@{#16981}
2017-03-02 20:33:50 +00:00
..
api
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
audio
Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream.
2017-03-02 01:02:23 +00:00
base
ArrayView: Support compile-time constant sizes
2017-03-02 20:33:50 +00:00
call
Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream.
2017-03-02 01:02:23 +00:00
common_audio
Enable GN check for webrtc/common_audio
2017-03-01 15:07:10 +00:00
common_video
Revert of H264BitstreamParser: Log nalu type when parsing slice type fails. (patchset #1 id:60001 of https://codereview.webrtc.org/2726833005/ )
2017-03-02 14:55:07 +00:00
examples
iOS AppRTCMobile: Fix SocketRocket bug
2017-02-27 13:20:18 +00:00
logging
Remove unused include from rtc_event_log_parser.cc
2017-02-28 09:46:19 +00:00
media
Pick a matching CN codec, rather than the first CN codec.
2017-03-02 19:03:25 +00:00
modules
Revert of Add QP for FFmpeg H264 decoder. (patchset #4 id:200001 of https://codereview.webrtc.org/2649133007/ )
2017-03-02 15:22:26 +00:00
ortc
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
p2p
Relanding: Remove the IceTransportInternal2.
2017-03-02 06:47:18 +00:00
pc
Support GCM ciphers even if ENABLE_EXTERNAL_AUTH is defined.
2017-03-01 23:34:36 +00:00
sdk
Android HW decoder: Limit frame size based on cropping information
2017-03-02 08:55:32 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Provide an inline implementation of IsEnabled.
2017-02-28 17:26:22 +00:00
test
Revert of fixed VP8 simulcast to not decode non-selected streams (patchset #5 id:80001 of https://codereview.webrtc.org/2728553003/ )
2017-03-02 12:59:33 +00:00
tools
Add probe logging to RtcEventLog.
2017-02-27 10:18:46 +00:00
video
Increase kMinRequiredSamples (5 -> 200) for updating histogram stats from OnFrameBufferTimingsUpdated callback.
2017-03-02 16:25:46 +00:00
voice_engine
VoE Utility: Fix a naming nit in RemixAndResample
2017-03-02 12:10:57 +00:00
.gitignore
…
BUILD.gn
Revert of Removing unused audio device tests (patchset #1 id:1 of https://codereview.webrtc.org/2716843002/ )
2017-02-26 15:12:50 +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
Move iOS tests to XCTest from gtest.
2017-02-14 12:58:56 +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.