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stefan 7aa9b910ff Fix issue with deprecated CongestionController interface not working. 
BUG=b/33446014

Review-Url: https://codereview.webrtc.org/2565503002
Cr-Commit-Position: refs/heads/master@{#15491}
2016-12-08 20:21:48 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Generalize FlexfecReceiveStream::Config.
2016-12-08 12:18:05 +00:00
audio
Refactor webrtc/{api,audio} and modules/audio_coding for GN check
2016-12-07 16:23:35 +00:00
base
Clean up redundant include of ../webrtc_overrides
2016-12-07 11:30:50 +00:00
build
Move /webrtc/api/android files to /webrtc/sdk/android
2016-12-06 12:29:45 +00:00
call
Generalize FlexfecReceiveStream::Config.
2016-12-08 12:18:05 +00:00
common_audio
Split targets mixing .c and .cc sources.
2016-12-06 06:47:52 +00:00
common_video
Sanity check parsed QP values from H264 bitstream
2016-12-02 09:29:53 +00:00
examples
Add FlexFEC settings toggle in Android AppRTCMobile.
2016-12-08 14:51:35 +00:00
libjingle
Remove all references to GYP
2016-11-16 19:11:38 +00:00
logging
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
media
Reland of Add ability to scale to arbitrary factors (patchset #1 id:1 of https://codereview.webrtc.org/2557323002/ )
2016-12-08 16:04:58 +00:00
modules
Fix issue with deprecated CongestionController interface not working.
2016-12-08 20:21:48 +00:00
p2p
Relanding: Refactoring that removes P2PTransport and DtlsTransport classes.
2016-12-07 00:22:11 +00:00
pc
Relanding: Refactoring that removes P2PTransport and DtlsTransport classes.
2016-12-07 00:22:11 +00:00
sdk
Reland of Add ability to scale to arbitrary factors (patchset #1 id:1 of https://codereview.webrtc.org/2557323002/ )
2016-12-08 16:04:58 +00:00
stats
Collecting RTCIceCandidatePairStats.transport_id and improved unittests.
2016-11-30 09:50:36 +00:00
system_wrappers
Add comment to metrics.h
2016-12-06 11:59:08 +00:00
test
Generalize FlexfecReceiveStream::Config.
2016-12-08 12:18:05 +00:00
tools
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
video
Rename RtpStreamReceiver::IsFecEnabled to RtpStreamReceiver::IsUlpfecEnabled.
2016-12-08 16:25:51 +00:00
voice_engine
Move WEBRTC_VOICE_ENGINE_TYPING_DETECTION to transmit_mixer.h
2016-12-05 10:22:18 +00:00
.gitignore
…
BUILD.gn
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +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
Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabled.
2016-12-02 16:58:02 +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
…
transport.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
typedefs.h
…
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 Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ )
2016-11-11 11:55:19 +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
voice_engine_configurations.h
Move WEBRTC_VOICE_ENGINE_TYPING_DETECTION to transmit_mixer.h
2016-12-05 10:22:18 +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.