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palmkvist 04055e95bf Removes all uses of FileWrapper in audio_device. 
BUG=webrtc:6463

Review-Url: https://codereview.webrtc.org/2386963003
Cr-Commit-Position: refs/heads/master@{#14811}
2016-10-28 09:08:29 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Testing of FileVideoCapturer.
2016-10-27 14:26:44 +00:00
audio
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +00:00
base
Fix "IsLoopbackIp" to cover all loopback addresses; not just 127.0.0.1.
2016-10-28 01:30:28 +00:00
build
Restore symbol level for Android builds.
2016-10-27 07:00:20 +00:00
call
Remove use of VoECodec in video/call tests.
2016-10-27 07:23:14 +00:00
common_audio
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
2016-10-17 06:56:20 +00:00
common_video
Reland of Move bitstream parser to more appropriate directory. (patchset #1 id:1 of https://codereview.webrtc.org/2430353004/ )
2016-10-26 09:48:24 +00:00
examples
Testing of FileVideoCapturer.
2016-10-27 14:26:44 +00:00
libjingle
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
2016-10-17 06:56:20 +00:00
logging
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +00:00
media
Implement framesDecoded stat in video receive ssrc stats.
2016-10-26 14:09:29 +00:00
modules
Removes all uses of FileWrapper in audio_device.
2016-10-28 09:08:29 +00:00
p2p
Fix "IsLoopbackIp" to cover all loopback addresses; not just 127.0.0.1.
2016-10-28 01:30:28 +00:00
pc
Introduce rtc::PacketTransportInterface and let cricket::TransportChannel inherit.
2016-10-25 17:50:41 +00:00
sdk
iOS: Optimize video scaling and cropping
2016-10-20 10:34:32 +00:00
stats
RTCStats equality operator added.
2016-10-25 11:31:27 +00:00
system_wrappers
Delete DataLog abstraction, which was almost unused.
2016-10-21 08:53:01 +00:00
test
Remove use of VoECodec in video/call tests.
2016-10-27 07:23:14 +00:00
tools
Use bayesian estimate of acked bitrate.
2016-10-27 15:19:34 +00:00
video
Remove use of VoECodec in video/call tests.
2016-10-27 07:23:14 +00:00
voice_engine
New statistics interface for APM
2016-10-28 08:32:24 +00:00
.gitignore
…
BUILD.gn
Remove remnants of libsrtp1
2016-10-25 16:47:31 +00:00
call.h
Integrate FlexfecReceiveStream with Call.
2016-10-24 06:37:24 +00:00
codereview.settings
…
common_types.cc
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
common_types.h
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
common.gyp
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
config.cc
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +00:00
config.h
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +00:00
DEPS
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete talk directory, and references from build_ios_libs.sh.
2016-09-02 10:07:13 +00:00
OWNERS
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
supplement.gypi
…
transport.h
…
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
video_encoder.h
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
video_frame.h
Reland of Delete webrtc::VideoFrame::CopyFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2397943003/ )
2016-10-24 07:44:17 +00:00
video_receive_stream.h
Implement framesDecoded stat in video receive ssrc stats.
2016-10-26 14:09:29 +00:00
video_send_stream.h
Implement framesEncoded stat in video send ssrc stats.
2016-10-24 09:09:39 +00:00
voice_engine_configurations.h
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
webrtc_examples.gyp
Reland of name AppRTCDemo on Android and iOS to AppRTCMobile (patchset #1 id:1 of https://codereview.webrtc.org/2358133003/ )
2016-09-27 08:10:52 +00:00
webrtc_tests.gypi
GYP: Remove targets inside include_tests==1 that are converted to GN.
2016-09-15 11:57:39 +00:00
webrtc.gyp
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +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.