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brandtr 8b5c345ee5 Add GUARDED_BY's in FlexfecReceiver. 
BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2589583004
Cr-Commit-Position: refs/heads/master@{#15688}
2016-12-19 18:02:30 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Removed undefined method from webrtcsession.h.
2016-12-19 17:34:14 +00:00
audio
Add ossu@ to OWNERS of audio/ and modules/audio_coding/
2016-12-15 15:52:14 +00:00
base
Revert of Delete unused code from systeminfo. (patchset #3 id:40001 of https://codereview.webrtc.org/2578323005/ )
2016-12-17 00:48:28 +00:00
build
Move tools/mb -> tools-webrtc/mb
2016-12-18 21:14:50 +00:00
call
Move FlexfecReceiveStream from api/call/ to call/.
2016-12-19 09:13:46 +00:00
common_audio
Fix for left shift of negative value in NetEq.
2016-12-14 10:00:02 +00:00
common_video
Reland of Delete VideoFrame default constructor, and IsZeroSize method. (patchset #1 id:1 of https://codereview.webrtc.org/2574123002/ )
2016-12-15 14:30:00 +00:00
examples
Add QP stats to the statsview in AppRTCMobile for ios.
2016-12-19 10:14:08 +00:00
libjingle
Remove extra uses of basictypes.h.
2016-12-09 14:54:08 +00:00
logging
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
media
Clean up storage of FlexFEC payload type in webrtc::VideoCodecSettings.
2016-12-19 17:41:04 +00:00
modules
Add GUARDED_BY's in FlexfecReceiver.
2016-12-19 18:02:30 +00:00
p2p
RTCStatsCollector: Utilize network thread to minimize thread hops.
2016-12-19 12:58:02 +00:00
pc
Removing "crypto_required" from MediaContentDescription.
2016-12-13 19:29:16 +00:00
sdk
Add disabled certificate check support to IceServer PeerConnection API.
2016-12-19 12:10:30 +00:00
stats
Rename RTCOutboundRTPStreamStats *_rtt members to *_round_trip_time.
2016-12-15 14:17:15 +00:00
system_wrappers
Revert of Delete unused code from systeminfo. (patchset #3 id:40001 of https://codereview.webrtc.org/2578323005/ )
2016-12-17 00:48:28 +00:00
test
Add multithreaded fake encoder and corresponding FlexFEC VideoSendStreamTest.
2016-12-19 13:47:28 +00:00
tools
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
video
Revert of Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame. (patchset #1 id:1 of https://codereview.webrtc.org/2574943003/ )
2016-12-19 17:36:33 +00:00
voice_engine
Wire-up audio BWE with overhead.
2016-12-15 14:24:52 +00:00
.gitignore
…
BUILD.gn
Re-enable Opus complexity tests on Android
2016-12-19 14:18:35 +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
CodecInst operator<<
2016-12-19 14:04:04 +00:00
config.cc
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
config.h
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
DEPS
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +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
Remove extra uses of basictypes.h.
2016-12-09 14:54:08 +00:00
video_decoder.h
Move VideoDecoder::Create() logic to separate internal video decoder factory
2016-12-01 08:27:35 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Reland of Properly report number of quality downscales in stats. (patchset #1 id:1 of https://codereview.webrtc.org/2586783003/ )
2016-12-19 14:32:16 +00:00
video_receive_stream.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
video_send_stream.h
Implement periodic bandwidth probing in application-limited region.
2016-11-28 21:11:24 +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.